IPitomy Wiki - User contributions [en]

IPitomy Cloud

2025-11-19T14:37:53Z

Tyler: Created page with "{{Short description|Advanced SIP softphone for the IPitomy Cloud PBX with BLF, advanced call controls, and multi-profile audio handling}} '''IPitomy Acrobits Desktop Phone'''..."

{{Short description|Advanced SIP softphone for the IPitomy Cloud PBX with BLF, advanced call controls, and multi-profile audio handling}}

'''IPitomy Acrobits Desktop Phone''' is a SIP-based softphone client designed for the IPitomy Cloud PBX. It provides VoIP calling, presence monitoring, call transfer, call parking, diagnostic logging, and configurable audio routing. Built on the Acrobits softphone engine, the application supports dynamic codec negotiation, NAT traversal, SRTP encryption (when enabled), and multi-account SIP operation depending on provisioning.

The application is used in call centers, offices, distributed workforces, and IPitomy reseller environments.

== Overview ==
The IPitomy Acrobits Desktop Phone connects to the IPitomy Cloud PBX via SIP registration using the user's extension credentials. Once registered, the software synchronizes presence data, directory listings, call history, and provisioning settings.

The interface consists of:
* A navigation panel (Keypad, Contacts, History, Settings)
* A main workspace (active module display)
* Presence indicators (BLF subscription results)
* Call control overlays (during active calls)

The application supports both IPv4 and IPv6 where configured by the PBX.

== SIP Architecture ==
The softphone registers to the PBX using standard SIP REGISTER messaging. It maintains the connection using periodic re-registration and SIP OPTIONS probing depending on PBX configuration.

Features include:
* SIP over UDP or TCP (provider dependent)
* SRTP capable (if PBX enforces or allows it)
* NAT traversal using STUN and rport parameters
* Adaptive jitter buffering
* Codec negotiation with priority list provisioning

Supported codecs typically include:
* G.711u (PCMU)
* G.711a (PCMA)
* G.722 HD Voice
* Opus (if enabled)
* GSM (fallback)

== Interface Modules ==

=== Keypad ===
The keypad provides:
* Manual number entry
* DTMF for in-call menu navigation
* On-screen call controls
* Quick Dial presence list

During calls, keypad usage sends DTMF tones using RFC 2833 (out-of-band) signaling. In-band tones may be used depending on codec and PBX preference.

=== Quick Dial and BLF Monitoring ===
Quick Dial entries can be configured with Busy Lamp Field (BLF) monitoring using SIP SUBSCRIBE/NOTIFY mechanisms.

Presence states:
* '''Available''' (SIP NOTIFY with "idle")
* '''Busy''' (call active or ringing)
* '''Unavailable''' (extension unregistered)
* '''Unknown''' (no subscription response)

BLF allows:
* Quick calling
* Attended or blind transfer to monitored extensions
* Monitoring parked call slots
* Supervisory observation in call centers

BLF presence is polled periodically or pushed through NOTIFY packets depending on PBX behavior.

== Contacts ==
The contacts module includes:
* System directory (retrieved from PBX provisioning)
* User-created Quick Dial entries
* Extensions, departments, and general contacts

Search behavior supports substring matching on:
* Name
* Extension
* URI phone number

== Call History ==
The application stores:
* Missed calls
* Answered calls
* Outgoing calls
* Calls answered on another device registered to the same extension

Each call record includes:
* Caller ID and name
* Timestamp
* Call duration (when applicable)
* Completed or canceled call state
* Ability to redial immediately

Call history is stored locally and may sync from server depending on PBX configuration.

== Call Handling ==
The application supports:
* Outbound calling
* Voicemail server access
* Multi-stage dialing (international)
* In-call switching of audio devices
* Call merging (if PBX conferencing enabled)

=== In-Call Controls ===

==== Mute ====
Disables sending audio via the microphone while maintaining RTP reception.

==== Hold ====
Places the remote party on server-based hold using SIP RE-INVITE.
The PBX plays Music On Hold (MOH), not the softphone.

==== Audio Switching ====
Users may switch:
* Microphone device
* Playback device
* Ringtone device

Switching creates a new local audio device binding without interrupting the SIP session.

== Call Transfers ==

=== Blind Transfer ===
Blind transfers use SIP REFER immediately without first calling the target.

Procedure:
# In an active call, select '''Transfer'''.
# Enter extension or number.
# Select '''Blind Transfer'''.
# Softphone sends REFER with target URI.
# Remote party connects directly to recipient.

Blind transfers fail if:
* The destination does not exist
* PBX blocks the transfer
* Target extension is forbidden by Class of Service

If a blind transfer fails, the original call may return depending on PBX settings.

=== Attended Transfer ===
Attended transfers involve placing the caller on hold and calling the intended recipient first.

Procedure:
# During call, select '''Transfer'''.
# Enter destination.
# Select '''Attended Transfer'''.
# Caller placed on PBX hold.
# Softphone initiates second SIP INVITE to contact.
# If recipient answers:
## User announces caller.
## User selects '''Complete Transfer''' (REFER or REFER via Re-INVITE).
# If recipient declines:
## User selects '''Return to Call'''.

If network or signaling interrupts the second call, the user may still retrieve the original caller.

=== Semi-Attended Transfer ===
If a user initiates an attended transfer but completes it before speaking to the recipient, it becomes a semi-attended (ringing) transfer.

== Call Parking ==
Parking allows callers to be placed into shared PBX holding slots.

Typical park slots:
* '''701'''
* '''702'''
* '''703'''

The softphone monitors these slots via BLF.

=== Parking a Call ===
# Place the call on hold.
# Dial park slot (e.g., 701).
# Hang up or complete transfer.
# BLF indicator for park slot becomes active.

=== Retrieving a Parked Call ===
Users can retrieve by:
* Clicking the BLF Park slot in Quick Dial
* Manually dialing the slot number

If a slot times out, the PBX typically rings the original extension back.

== Settings ==

=== About ===
Displays:
* Version, build number
* Platform info
* Provisioning metadata
* Licensing information

=== Account Setup ===
Fields include:
* SIP username (extension ID)
* SIP password
* Registrar server
* Outbound proxy (if provisioned)

=== Notifications ===
Allows selection of:
* Ringtone tone
* Text alert sound
* Quiet modes (OS-dependent)

=== Sound ===
Sound configuration supports:
* Device enumeration for microphones and speakers
* Gain control (software amplification)
* Ringtone device independent of call audio
* Keypad tone output
* Outgoing noise suppression using DSP
* "Mute other applications" uses OS-level audio session attenuation

RTP audio stream analysis adjusts jitter buffer sizes dynamically.

=== Call Recording ===
If enabled, softphone recordings:
* Are stored locally
* Use raw PCM or compressed format depending on engine
* May synchronize with PBX policies
* Must follow local compliance laws (user must verify)

=== Controls ===
Options include:
* Launch at login
* Incoming call alerts (Full, Minimal, Notification-only)
* Default calling app registration
* Always-on-top mode
* Setup wizard for initial permissions

=== Troubleshooting ===
This tab includes advanced diagnostic functions.

==== SIP Log ====
Shows SIP registration and messaging:
* REGISTER
* INVITE/BYE
* REFER
* SUBSCRIBE/NOTIFY
* OPTIONS keep-alives

Developers and admins use logs to analyze:
* Registration errors
* NAT traversal behavior
* Codec negotiation
* Transfer issues

==== Diagnostic Data ====
Collects:
* Pre-DSP mic audio
* Post-DSP mic audio
* Playback processing stages

==== Problem Reports ====
Exports:
* SIP logs
* Device info
* OS audio routing data
* Crash logs (if present)

== Logout and Reset ==
'''Logout''' unregisters SIP credentials.

'''Reset Application''' returns all settings to defaults by wiping:
* Cached SIP credentials
* Audio preferences
* Contact lists
* Provisioning files

== Troubleshooting ==
Common issues and behaviors:

=== One-Way Audio ===
Typically caused by:
* NAT firewall blocking RTP
* Wrong audio device bound
* VPN routing SIP incorrectly

=== Registration Timeouts ===
Caused by:
* Incorrect credentials
* ISP SIP ALG interference
* PBX unreachable

=== BLF Not Updating ===
Caused by:
* PBX disabling presence
* Network packet loss blocking NOTIFY

== Network Requirements ==
Recommended:
* 100 kbps per call (G.711)
* QoS prioritization (DSCP EF)
* Disable SIP ALG on routers
* Stable latency under 150ms

== Security ==
Compatible with:
* SRTP (AES-128)
* TLS SIP signaling
* PBX-side authentication policies

Users may enforce:
* Strong passwords
* Encrypted transport
* Limited IP ACLs on the PBX

== Support ==
Support is provided by IPitomy Communications.

* Phone: 1-800-IPitomy
* Email: support@ipitomy.com
* Website: https://www.ipitomy.com

IP PBX Manual Branch Offices

2024-08-13T20:29:59Z

Tyler:

__NOTITLE__ {{IP_PBX_Manual|sortkey=Branch Offices}}

== Branch Offices ==

Branch Offices provide a powerful tool for interconnecting PBX Systems. Through Branch Offices you can create a network of systems where dialing phones on remote systems is as easy as dialing local extensions. You can route calls inbound to a destination on any branched PBX, and outbound through the trunks of other systems networked in this way. Branch office extensions can even participate in ring groups.
[[File:Samplebranchoffice.png]]

{| style="border-spacing:0"
|-
| style="border-top:0.0069in solid #000000; border-bottom:0.0069in solid #000000; border-left:0.0069in solid #000000; border-right:none; padding-top:0in; padding-bottom:0in; padding-left:0.075in; padding-right:0.075in" |
| style="border:0.0069in solid #000000; padding-top:0in; padding-bottom:0in; padding-left:0.075in; padding-right:0.075in" | '''IMPORTANT: Branch Offices communicate using IAX2 protocol over Port 4569. Refer to the Port Forwarding Configuration Table earlier in this manual for more information.'''
|}
[[File:editbranchoffice.png]]
{| style="border-spacing: 0px" class="wikitable"
|-
| <center>'''Sections/Fields'''</center>
| <center>'''Description'''</center>
|-
| style="text-align: center" | '''Name '''
| The name that is associated with the branch office. '''<big>Both sites involved in the Branch Office connection must use the same Name.</big>'''
NOTE: This field will not allow spaces in the name. It must be all one word. Symbols can be used (i.e. underscore “_” dashes “-”, etc.) can be used in place of spaces.

|-
| style="text-align: center" | '''Host '''
| Enter the static IP of the system your PBX is connecting to. If they do not have a static IP address, enter the word dynamic.
|-
| style="text-align: center" | '''Dial Prefix '''
| The dial prefix must be an asterisk, plus two numbers (ie. *22). This prefix must be dialed to reach any destination on another PBX that has not been entered as a Branch Extension.
|-
| style="text-align: center" | '''Password '''
| This is the password that will allow access to the PBX. Both sites involved in the Branch Office connection must use the same Password.
|-
| style="text-align: center" | '''Register '''
| YES - Only used if the Host is set to dynamic.
NO - does not require that the branch office register with the PBX system.

|-
| style="text-align: center" |''' Enable As Trunk '''
| YES - Branched PBXs are able to dial calls out your PBXs trunks.
NO - Branched PBXs are not able to dial calls out your PBXs trunks.
|-
|style="text-align: center" | '''Trunk Settings'''
|Check to Generate Ringing, Allow Call Recording, Allow CID Override, or Allow Outbound Transfer. Also allows you to set CID Override if this trunk is used in a branch connection.
|-
| style="text-align: center" | '''Enable Multisite Manager Connection '''
| Defines if Multi-Site Call Manager functionality is enabled. This feature requires a license from IPitomy to be functional.
|-
| style="text-align: center" | '''Qualify '''
| The time in milliseconds that the PBX should check that the branch office is still online. Setting this to zero may increase Branch Office stability.
|-
| style="text-align: center" | '''Enable IAX2 Trunking '''
| Indicates whether to use IAX trunking. If enabled (YES), this setting can help eliminate packet overhead by cutting the cost of continuous communication. 
|-
| style="text-align: center" | '''Allow Codecs '''
| Defines what codecs this branch office can use for calls. Priority for codecs are defined from top to bottom.
|-
| style="text-align: center" | '''Class of Service '''
| Allows for individual Branch Offices to use different COS. Setting to None will follow the default Branch Office COS defined under PBX SetupGeneral. 
|}
[[File:branchofficepage.png]]
===Configuring Office 1===

'''STEPS:'''

#Click on '''Destinations->Branch Offices'''. The '''Branch Offices''' page is displayed.
#Click on the '''ADD office''' button. The '''Edit Branch Office''' page appears.
#Give a unique '''NAME''' for the connection. The name should only contain alpha-numeric characters and no spaces. It should be '''ONE WORD ONLY'''.
#Enter the '''External IP''' of the PBX in the '''Host''' field.
#Enter a unique '''DIALING PREFIX''' of an asterisk followed by two numbers.
#Give a unique '''PASSWORD''' for the connection.
#Select '''NO''' for '''REGISTER '''(Note: that registration is not required if host is known).
#Configure '''Enable Trunking '''as needed.
#Configure '''Enable Multisite Manager Connection''' as needed.
#Set '''QUALIFY''' at '''0'''.
#Configure '''Enable IAX2 Trunking''' as needed.
#Click the [[File:savechanges.png]] button to save the changes.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

===Configuring Office 2===

'''STEPS:'''

#Click on '''Destinations->Branch Offices'''. The '''Branch Offices''' page is displayed.
#Click on the '''ADD Office''' button. The '''Edit Branch Office''' page appears.
#Enter a name in the '''NAME''' field that matches that given to the Office 1 PBX. The name should only contain alpha-numeric characters and no spaces. It should be '''ONE WORD ONLY'''.
#Enter an '''external''' '''IP address''' or '''domain''' corresponding to the IP of the main office PBX in the '''HOST''' field.
#Enter a '''UNIQUE''' dialing prefix for the extensions connected to the Office 2 PBX.
#The '''PASSWORD''' for Branch Office 2 needs to be the '''same''' as the one assigned in the Branch Office 1 PBX.
#Select '''NO''' for '''REGISTER'''.
#Configure '''Enable Trunking '''as needed.
#Configure '''Enable Multisite Manager Connection''' as needed.
#Set '''QUALIFY''' at '''0'''.
#Configure '''Enable IAX2 Trunking''' as needed.
#Click the [[File:savechanges.png]] button to save the changes.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

To setup additional Branch Offices in the PBX, use the steps outlined above. Keep in mind to use the same logic (pattern) and make sure that the dialing prefix is unique to each office.

=== Edit Branch Office ===

'''STEPS:'''

#From the '''Destinations->Branch Office '''page, locate the schedule name that you want to edit.
#Click on [[File:penciledit.png]] icon to the right of the '''Name '''of the Branch Office you want to update. The '''Edit Branch Office '''page appears.
#Edit the necessary parameters to configure the branch office.
#Click the [[File:Savechanges.png]] button to save the changes.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

=== Delete Branch Office ===

'''STEPS:'''

#From the '''Destinations->Branch Offices '''page, locate to the '''Name''' from the Branch Office listing that you want to remove.
#Click on [[File:deleteselected.png]] icon to the right of the '''Name '''of the '''Branch Office''' you want to delete. The Branch Office is removed from the listing page.
#Click the [[File:savechanges.png]] button to save the changes.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

=== '''Branch Extensions''' ===

Branch Extensions are created through the add extension field at the bottom of the branch office edit page. Branch extensions can be dialed directly without a prefix, provided that these extensions are properly configured on the PBX for which the branch extensions are defined. Branch extensions will appear in call routing drop down lists throughout the system after they are created. If needed, you can add Ring Group and Menu numbers as Branch Extensions.

==== Configuring Office 2 with Branch Extensions ====
[[File:branchextensions.png]]
'''STEPS:'''

#Click on '''Destinations->Branch Offices. '''The Branch Offices page appears.
#Click on the '''Branch Office''' connection ('''Name''') assigned to Office 1.
#Assuming Office 1 has extension numbers 100 thru 110, enter the numbers 100 to 110 in the field above the Add button then click '''ADD'''. You will enter (add) each extension one at a time. '''Note: Using the format X-Y will add all extensions in the specified range.'''
#If the extension number is valid (not already in use), the new extension will appear in the list of Branch Extensions.
#Click the [[File:Savechanges.png]] button to save the changes.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database.
#Verify the new extension by calling the number from phones located (registered) at Branch Office 2.

To place a call from one Branch Office to another simply dial the prefix that was assigned to that locations PBX + the extension of the user trying to be reached at the other office. This same easy concept works for transferring calls from one Branch Office to another. Or, if you have the extension configured as a Branch Extension on your PBX, you can simply dial the extension number and it will be automatically routed to the branch.

==== View Branch Office Extensions ====
[[File:Showextensionspage.png]]
 
'''STEPS:'''

#From the '''Destinations'''->'''Branch Office '''page, click on the '''Show Extensions''' button located at the top of the Branch Office list page.
#A popup window appears displaying all the extension information for each Branch Office that is currently in the system.
#Click on the “'''X'''” to close the popup window.

==== Parking at a Branch ====
In order to Park and retrieve parked calls at a Branch Office, you will have to do a little bit of programming.

'''Parking'''
*Create a Phantom Extension at the remote end PBX
*Add this as a Branch Extension in the local PBX
*Unconditionally Forward this ext to PSTN number ##700#
*When a user wants to park a call at the Branch Office, have them transfer the call to the Branch Extension

'''Retrieving Park'''
*Note the Branch Prefix Code
*To retrieve a call Parked at a branch, dial the prefix code plus the slot the call is parked in
**eg. With a Branch Prefix Code of *28 and a call Parked in 702, the user will dial *28702 to retrieve the call

Q Manager

2024-07-16T17:48:20Z

Tyler:

{{IP_PBX_Manual|sortkey=QManager}}

Q Manager is a Windows application that allows users to monitor and interact with calls. Your PBX must be on version 4.6.1 in order to use Q Manager, and 4.8.0+ to use the integrated chat feature. Licenses are sold to allow extensions access to the Q Manager feature.

NOTE: Currently QManager can only retrieve calls parked via the standard Park feature, not Directed Park.

Right click, and select Save Link to download QManager [[https://www.ipitomy.com/pbx_files/qm3/QManagerSetup.exe QManager Download]]

=== NON-Windows PC Users / Web Manager ===

If you have a user that has MAC and/or not a Windows PC, Q Manager software will not run. To accommodate this, we have created something we call Web manager. To access, the user will still need to have a license assigned to their extension for Q Manager, but they will access a webpage instead of running a piece of software. The address for Web Manager would be <PBXipAddress>/ippbx/WebManager. The user will be prompted to login, the username will be their extension number, and the password will be their voicemail PIN. Web Manager does not have all of the advanced functionality of Q Manager, but you are able to left click on a call and do some actions.

NOTE: The URL is case sensitive.

*Pickup: Use this to grab a call that is ringing on an extension other than yours.
*Page: Use this to page another extension.
*Call: Use this to call another extension.

Idle extensions will show as Green, Ringing extensions will show as Light Green, and extensions on a call will show as Red.

Parked Calls will show on the right side of the screen, clicking a parked call will bring up an option to Pickup, which retrieves that call from park. (NOTE: Currently this will not work with Directed Park, only the standard park feature)

[[File:WebManager.jpg|File:WebManager.jpg]]

== Installation ==

*Log into the PBX and assign Q Manager licenses to extensions. This is found under PBX Setup=>Services.
**User licenses have basic functionality, QManager licenses gain the ability to Barge and to Drag/Drop calls from Queue onto particular extensions. Agent and Operator are not used at this time and are there for future possible functionality.
*Save and Apply Changes.
*Click the Restart Call Manager button.
*Q Manager requires that your PC have .NET Framework 4.0 installed.
*Once .NET Framework 4.0 is installed, right click, and select save link to download QManager [https://www.ipitomy.com/pbx_files/qm3/QManagerSetup.exe QManager]
**NOTE: You may need to be using Internet Explorer to one click install.
*Click Settings in Q Manager if the settings are not displaying on launch
*Set Username to an Extension Number that is assigned a license
*Set Password to the Voicemail PIN of the Extension
*Set Server Address to the IP of the PBX
*Leave Port at the default of 5048
*Save and so long as the PC is in an IP range included in the ACL for Call Manager, you will have a working copy of Q Manager.

[[File:QManager-GeneralSettings.jpg|File:QManager-GeneralSettings.jpg]]

*NOTE: If you would like to use Q Manager as a remote user, we advise that the remote PC come in over a VPN.

==Training Video==
The following [[http://www.youtube.com/watch?v=ArDyehd6dXI| Video]] goes over how to use Q Manager, with some basic setup instructions near the end.

[[http://wiki.ipitomy.com/wiki/Q_Manager_Icons| Q Manager Icons Explained]]

== Chat ==

The PBX has a built in XMPP Server that allows Q Manager to use Chat.

*Under PBX Setup=>Chat, make sure Enable Chat Server is set to Yes
*Ensure that Chat Server Status says Running
*Apply Changes

[[File:QManager-ChatConfig.jpg|File:QManager-ChatConfig.jpg]]

*Click Settings in Q Manager
*Click Chat and ensure its enabled
**Q Manager automatically uses the same settings for the extension

NOTE: Chat must be enabled for a user to see an Abandoned Call Notification for a ring group.

[[File:QManager-ChatConfig2.jpg|File:QManager-ChatConfig2.jpg]]

IPitomy Fax

2024-03-15T15:30:48Z

Tyler:

__TOC__

== IPitomy E-Fax ==
IPitomy Fax Services offers a virtual fax solution for your client by Email to Fax, Web to Fax, Print to Fax, or Mobile E-Mail to Fax.

[[File:How Fax Works.png|alt=|File:FaxOverview.jpg]]

== Email to Fax ==

::NOTE:: No technical fax size limit on our end, but the email provider may have a max attachment size. It is advised to keep faxes under 100 pages<gallery mode="slideshow" widths="240" heights="240">
File:EmailtoFax (1).JPG
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</gallery>

Email to Fax must be sent to: destinationnumber@ipfax.net 2) Users MUST send from the email address they are registered to on the system as the system utilizes the ‘from’ email address as their Username or Login for authorization. Clients MUST be registered in the system with their email address as their Login/Username. 3) The ‘Subject’ line of the email MUST include the word ‘pass’ followed by the User’s password 4) Email to Fax is recommended be sent in Plain Text format 5) Clients may attach up to three attachments for faxing. Almost all attachment formats are supported. 6) Anything in the body of the email will be included in the cover page of the fax. An empty body will result in no cover page being sent and only the attachment(s) being faxed. 7) Clients may include the fax recipient’s name (on cover page) by including it as the first words in the ‘Subject’ field of the email. 8) Clients may include a subject for the fax by including ‘s=subject’ in the ‘Subject’ field of the email for faxing. (the word subject to be replaced by actual subject) Example of addressing of email for faxing

== Web to Fax ==
:When registering for IPitomy Fax you will receive an email with your login information. Save this email/information.
::NOTE:: Max of 3 file attachments per fax, max 2MB size per file.
<gallery widths="220" heights="220" mode="slideshow" caption="IPitomy Web to Fax">
File:Web2Fax (1).JPG
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File:Web2Fax (11).JPG
File:Web2Fax (12).JPG
File:Web2Fax (13).JPG
</gallery>To send a Web to Fax, a User will log into the User Portal [http://secure.ipfax.net http://secure.ipfax.net] and do the following: 1) Address who the fax is to be delivered to. (John Smith in Example) 2) Include a Fax Subject if desired. Fax Subject will be included on Cover Page of fax as well in Confirmation Report and Call Record for easy identification by the User 3) In the Fax Number(s) field, type in the fax number(s) the fax is destined for. Include country code (‘1’ for N. America but no prefix for international calls such as 011). Web to Fax can send the same fax to 10 destination numbers at the same time. 4) Upload up to three attachments. Almost all formats are supported. 5) Click Send Fax Now! . User will receive confirmation emails as configured on their account and may check Online Reports for real-time status.

== Print to Fax ==

=== Installation ===
Please Install the following program, for Windows PC's only.   MAC please use Web to Fax

IPitomy Print to Fax Installation [http://download.pangea-comm.com/ftp/printdrivers/InternetFax-v11.0.1-TLS-64bits-latest.zip Windows 64bit] [http://download.pangea-comm.com/ftp/printdrivers/InternetFax-v11.0.1-TLS-32bits-latest.zip Windows 32bit]

[[Installation Guides]]

=== Guide ===
<gallery mode="slideshow" widths="120" heights="120">
File:Print To Fax (1).JPG
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</gallery>

Training:Application Solution

2023-12-11T15:20:11Z

Tyler: /* The Application Development Sequence: */

= IPitomy Technical Training – Basic: Understanding the Application =

== Overview ==

 At IPitomy, we believe in crafting a PBX solution that precisely meets our customer's requirements. The key to achieving this is thorough information gathering from the customer. This information forms the basis of your IPitomy setup worksheet, a critical document defining the scope of work for system programming. Having this documented ensures alignment of expectations and serves as a reference for system reconstruction, if necessary.

= Database Building: =

==== Database Building in IPitomy's IP PBX System ====
At IPitomy, our IP PBX system represents the cutting edge of business telecommunications, functioning as a software-centric solution. This approach distinguishes it from traditional Time-Division Multiplexing (TDM) based key systems and PBXs. The primary advantage of our software-based system is its minimal reliance on hardware components, significantly reducing the likelihood of hardware failures.

Effective database construction within our system is crucial and should be meticulously planned. This planning process is vital to ensure that the final solution aligns perfectly with customer expectations. Gathering detailed information prior to the installation date is essential for a smooth, professional installation process and to minimize unbillable follow-up work.

One of the standout features of the IPitomy system is its user-friendly, web-based administration interface. This interface simplifies the programming and setup of the application. It's important to note that the IPitomy system does not include a default database. Therefore, setting up the database requires careful attention to sequence to establish the necessary elements for successful call flow.

==== Key Components of Database Building ====

# Destinations:
#* In the IPitomy system, destinations are the endpoints within the PBX where calls are routed. Examples include automated attendants or individual extensions.
#* Direct Inward Dialing (DID) numbers can be assigned to directly route calls to specific destinations, such as conference rooms or individual extensions, enhancing direct communication efficiency.
# Providers (Trunks):
#* Providers in the IPitomy system are crucial elements that manage the routing of calls both into and out of the system.
#* Once destinations are established, providers can be set up to facilitate the designed call routing requirements. This setup is integral to ensuring that calls are managed and directed according to specific business needs.

By following these steps and understanding the intricacies of our IPitomy IP PBX system, users can leverage its full potential to create a robust, efficient telecommunication setup that meets and exceeds customer expectations.
*

[[File:Menu Left detail.png|none|Menu Left detail.png]]

=== The Application Development Sequence: ===

==== Creating Extensions: ====

# Extension Configuration:
#* Extensions are configurable as 3 or 4-digit numbers.
#* Features include various Call Forwarding options such as Unconditional, Busy, No Answer, or Unavailable.
#* With the creation of an extension, a corresponding voicemail box, a chat client, and a schedule are automatically generated.
#* The 'Follow-Me' feature enhances flexibility by allowing simultaneous and sequential ringing across multiple extensions or PSTN numbers.
# Licensing and Extension Management:
#* The total number of available extensions is governed by the licensing agreement, with options for expansion to cater to growing business needs.
#* Separate licensing categories exist for IPitomy phones and non-IPitomy devices, with open extension licenses typically carrying a slightly higher cost.
#* The system provides Auto Provisioning for IPitomy Phones, streamlining the setup process.
#* Importing user information through a CSV file is a time-efficient method to batch-create extensions.
#* Manual creation of extensions is also an option, allowing for individualized data entry.
#* Mass editing features are available post-creation, facilitating easy management and updates of extension information.
# System Compatibility and Intercom Use:
#* The IPitomy system boasts wide compatibility with a range of SIP devices, including conference phones, softphones, and other SIP-compliant equipment.
#* An added feature is the capability of using IPitomy phones for intercom paging, either between individual extensions or groups, enhancing internal communication efficiency.

==== Creating Groups of Extensions: ====

# Group Functionality and Customization:
#* Groups can be configured to ring all extensions at once or in a specific sequence, depending on the organizational needs.
#* Advanced group functionalities offer remarkably flexible call coverage, accommodating diverse communication strategies.
#* Priority settings within groups enable automatic call assignment to specific members during peak times, bypassing the need for manual login.
#* Timeout settings can be programmed to redirect calls to alternative destinations after a set duration.
#* The agent ring time feature allows for the re-initiation of the ring sequence to include members who have just concluded other calls.
#* Groups typically function as hubs for routing incoming calls to a department or team.
#* Groups can also be utilized for paging purposes, with both unicast call paging and multicast paging options.
#* The system allows for customization of caller ID information, and unique Music on Hold or Message on Hold can be assigned to each ring group.

==== Creating Menus and Recording Prompts: ====

# Automated Attendant Configuration::
#* Menus in the IPitomy system serve as single-digit dialing destinations, simplifying the call routing process for callers
#* Users can record prompts using the in-built prompt recording utility, ensuring clarity and professionalism in automated responses.
#* Planning and pre-recording prompts are advisable for efficient onsite configuration.
#* The system also supports the uploading of pre-recorded prompts, providing flexibility in the creation and management of automated attendants.

==== Additional System Features & Destinations ====

# Conference Bridge Implementation:
#* The system includes default conference bridge destinations (901 and 902), each accommodating up to 32 participants.
#* Additional conference bridge destinations can be activated through license expansion, catering to larger or multiple simultaneous conferences.
#* Customizable greetings can be set for conference rooms, providing a personalized experience for participants.
# Integrated Voice Mail System:
#* Voice mail boxes are automatically created for each extension, streamlining the setup process.
#* The integrated nature of the voice mail system eliminates the need for manual forwarding, with default settings to forward calls to voice mail after 32 seconds of inactivity.
# Scheduling and Time-Based Routing:
#* The scheduling feature in the IPitomy system allows for the dynamic routing of calls based on the time of day, enhancing flexibility in call management.
#* Schedules can be customized to route calls differently during business hours, after hours, and during specific intervals like lunch breaks.
#* This feature can be implemented at various points in the call path to facilitate time-sensitive call routing.
# Branch Office Connectivity:
#* The branch office destination feature enables seamless call routing to other IPitomy IP PBX Systems.
#* Extensions within branch offices are easily accessible, maintaining organizational coherence across multiple locations.
#* A unique dialing code is available to connect extensions that share the same number in different branches.
# Scheduled Call Announcements:
#* This feature allows for the automated playing of announcements or audio files to a group of phones or integrated paging devices based on a preset schedule.
#* Practical applications include school bell sounds to mark the start and end of classes or emergency announcements triggered by dialing a specific extension and entering a PIN code.

== Review and Confirmation Process in IPitomy System Setup ==

The final stage in setting up an IPitomy IP PBX system is a comprehensive review and confirmation process. This ensures complete alignment of the system components with the customer's specifications and requirements. Documenting all relevant information in the IPitomy Setup Worksheet is crucial for defining the scope of work and establishing clear expectations between the provider and the customer.

==== Extension Number Range ====

* Confirm the range of extension numbers to be used, ensuring they fit the customer's organizational structure and communication needs.

==== Device Types ====

* Document the types of devices integrated into the system, including IP phones, conference devices, and other SIP-compatible hardware.

==== Network Layout ====

* Examine and confirm the overall network layout, assessing its compatibility and efficiency for the IPitomy system.

==== Router Specifications ====

* Ensure understanding of the router's capabilities, crucial for network settings configuration and system integration.

==== Data Switch Types ====

* Identify data switch types, particularly noting Power over Ethernet (PoE) capabilities, essential for planning device connectivity and power management.

==== Cabling ====

* Check that network cabling is certified and in good condition, and that all devices and IP addresses are accurately documented.

==== Router Access ====

* Secure access to the router GUI or establish a relationship with IT personnel for necessary access.

==== IP PBX IP Address ====

* Confirm the uniqueness of the IPitomy system's IP address within the network to avoid conflicts.

==== Network Condition ====

* Assess the overall health and performance of the network, ensuring it can adequately support the IPitomy system.

==== End-User Network Responsibility ====

* Ensure that the end-user understands their role in maintaining network performance, including the necessity of network upgrades or modifications for optimal system functioning.

*

== Application Development ==

=== Voice Mail Email ===

In order for the IP PBX System to send out emails, it is necessary to have an email account assigned to the system so all emails that the system sends out can be from a legitimate email account.  This is entered into the system under <PBX Setup> <Voicemail>.  See the screen below.

Once the email settings are properly configured for sending emails, all that is required is to add the users email address in the extension.  Omitting the email address turns off the email feature in each extension.  if you are not using Voice mail to Email, do not put an email address in the email address field on the extension screen.

Email needs to be setup for sending out of the PBX to the customer's email system. 

*Sent From
***The customer must provide an Email account for the Voice Mail system (vmail@company.com) on their Email Server and use that email account so all emails from the IP PBX System will be sent using the accounts email credentials.
***If no email server is available to create an account, creating one on Google GMail (vmail.company@gmail.com) or another similar service.  Be sure to prepare for this in advance and have an email account and password ready when you go to do the installation. If you neglect to do this, it will add to your installation time.

[[File:VM to email.PNG|center|VM to email.PNG]]

=== Ring Groups/ACD ===

*Ring Groups Defined – Ring Groups are a powerful communications resource in the IPitomy IP PBX and for your customer.ACD will be discussed in the Advanced training course.<ul style="list-style-type:circle;">
</ul>

*Ring Strategy
***Go to the IPitomy WIKI at [http://wiki.ipitomy.com http://wiki.ipitomy.com]
***Select the IPitomy PBX Plus Administration Guide
***Select Groups or search for - Ring Stategy  - The WIKI will have all of the information required to set up ring groups.

**Members – those in the group with physical telephones. Members are permanent devices in the Queue. Members have the ability to place their device into Pause thereby removing themselves from queued calls, however this also eliminates all incoming calls except Page Announce Call
**Agents – those in the group with no specific telephone or wanting the ability to Log On to the Queue and Log Off the Queue separately from telephone-based functions.  Agents can log in from any phone. - Requires ACD Option
**Failovers (Queue Timeout destinations) Where the call is directed to after the timeout expires.  This can be any destiniation in the system.
**Will a menu be associated to any Ring Groups (ACD Feature – callers in an ACD - Ring Group queue can interact with options available while waiting in queue and select a new system destination)
**Use theIPitomy WIKI at [http://wiki.ipitomy.com http://wiki.ipitomy.com] to search for any of the subjects above to learn about how to implement the powerful features of Ring Groups and ACD.
**Link to Group Video Training [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

=== Menus ===

*Menus defined

*Auto Attendant, and subsequent menus must be planned in advance and well organized to allow for a streamlined installation of that portion of the application.
***Get prompts scripts – write them or have them prepared for you by the user
***Get Destination selections for one-digit dialing
***Determine if extension dialing will be allowed and at what menus
***Determine Menu overflow Destinations – where callers will be routed when they dial:
****nothing
****incorrectly
***Link to Video Training [https://www.youtube.com/watch?v=XjRzyUEpOfI&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

=== Schedules ===

*Determine schedules
*Day Hours of operation
*Lunch Hour
*Night Hours of operation
*The “Attendant” assigned telephone may be given the ability to select the Day/Night mode of operation

=== Branch Office ===

*If a Branch Office is to be part of this application the numbering plan and menus must be known for that system and it must be deployed with equal attention to detail.

Determine:

*If extension numbering at the branch office is to be transparent to the users (users may dial any extension number regardless of branch location or local PBX location with no special coding or prefix erquired). If so, the extension number scheme at the branch office must not conflict with extension numbering at this PBX or any other branch office.
*Shared Name for the Branch – this name will be applied to both PBX locations to allow them to meet (connect) over the interlinking network (Internet usually)
*Code for branch access – this code can be mirrored at each branch and used to access the paired branch
*Branch Password – this password will be applied to both PBX locations to allow them to meet (connect) over the interlinking network (Internet usually)

=== Remote Phone(s) ===

*If remote phones are to be used, assure that the router has been programmed to allow BOTH UDP and TCP Packet forwarding (Port 5060 should be configured for UDP and Ports 10,000-20,000 should be configured for UDP)
*Assure that the bandwidth at the remote phone location is adequate to handle call traffic for each telephone – especially when multiple phones are deployed at remote locations. (Plan for 200kb/s for each, two-way voice call.)
*If more than five remote phones are to be used at any remote site concurrently, consider installing an IP PBX at that location as a Branch Office instead.
*QOS is essential to all lans with VoIP traffic. We recommend setting QOS if possible to make voice call traffic a higher priority than other data traffic.
*Disable ALG (Application Layer Gateway). This router function can be powerful but a nuisance to voice traffic.
*If a Sonicwall is in use, review notes in IPitomy’s Sonicwall Configuration guide at: [http://www.ipitomy.com/webrelease/Sonicwall/Sonicwall%20Quick%20Guide.pdf http://www.ipitomy.com/webrelease/Sonicwall/Sonicwall%20Quick%20Guide.pdf]
*Go to PBX Setup=>Phone Global - enable Phone Download Enabled     - enable Phone Auth Enabled     - note the user and pass to be manually entered later (since the phone is already remote) - Click Save and Apply Changes Edit the Phone settings for the extension (pencil with handset) - Change the Configuration Updates protocol to HTTP - Click Save & Configure Phone button (were the phone local, the correct values would be sent to the phone) Manually Changing in phone - Log into phone via IP Address (user: root, pass: root) - Navigate to Phone Maintenance=>Autoprovision - Change Protocol to HTTP - Enter Username and Password from earlier Phone Global steps - Change the Software Server URL to: [http://ippbx/phonecfg/ http:///ippbx/phonecfg/]
*- Click Submit and wait for the phone to become idle
* Refer to this section of the manual : [[HD Phones#Remote Phones|Remote_Phones]]

Below is how you include the remote phones section: 

=== [[Qmanager|QManager]] ===

*Desktop Call Manager is a PC-based, Windows Application that can be loaded onto user computers to gain a high level of control of communications for their telephone. DCM is licensed per user and can be installed at on a single PBX or multiple PBX’s that are branched together with the Multisite DCM License.
*Since Desktop Call Manager integrates a Chat Client, DCM links the desktop to the world of chat and SMS Texting.
*Presence – DCM enables a presence indication via its integrated Chat client.
**DCM Provides the user with:
**Ability to monitor selected extensions on the IP PBX and Branch Office IP PBX’s
**Monitor call traffic at the monitored extensions
**Interact with call traffic at the monitored extensions
****Listen
****Whisper
****Barge
****Record calls in progress at that extension (Recordings via DCM are stored in the DCM application)

**Interact with callers in voicemail
***Screen caller leaving messages in voice mail
***Pick up (retrieve) callers from voice mail
**Record calls in progress at their own extension (Recordings via DCM are stored in the DCM application)
**Monitor Park Locations
**Monitor Trunks
**Utilize DCM-based Speed Dial
**Send and Receive Text Messages (a Chat server is required – any may be used)
**Monitor Conference Rooms 901 902… and other if licensed/programmed
**DCM may be used to invoke PBX functions:
***Dial
***Transfer
***Park
***Hang up
***Call and Extension
***Page an Extension
***Call Forward – DCM can be used to monitor PC activity and invoke pre-programmed call forward scenarios when a PC user is inactive for 15 minutes

=== Multisite Desktop Call Manager ===

*Both sites must be Multi Site Licensed and Multi Site Enabled
*MUST be enabled in the Branch office settings
*You MUST Port Forwards Ports 5048 and 5038 to the IP of the PBX at each router
*You must set up the ACL in the IP PBX to allow the mated branch office to connect on Ports 5038 and 5048. (The default only allows for local IPs).

 Questions:

#'''T'''/F, A Branch Office extension can be dialed directly without a branch office code regardless of which extension and branch where the extension is installed.
#What benefit comes from setting up a Sent From email address. '''a. puts the Sent From email address in front of the user each time an email from the IP PBX is received.''' b. puts the Sent From email address in front of the user each time an email is received. c. The email address set as Sent From becomes the server for all emails that are IP PBX generated d. The Sent From email address defines which email that are received are valid.
#T/'''F''', When DCM is used at a branch office a Multisite license must be purchased at only the IP PBX that is the Branch Office IP PBX.
#'''T'''/F, A Menu can be assigned to a ACD Group so that callers may dial a single digit and while waiting in queue and be directed to another system destination.

Where on the WWW can you go to find the IPitomy Installation and Maintenance Manual? [http://wiki.ipitomy.com http://wiki.ipitomy.com]

or

[http://www.ipitomy.com/webrelease/IPitomy/IP1100+/IPitomy%20IP1100+%20Manual.pdf http://www.ipitomy.com/webrelease/IPitomy/IP1100+/IPitomy%20IP1100+%20Manual.pdf]
[[Category:Training]]

Training:Application Solution

2023-12-11T15:06:05Z

Tyler: /* Detailed Confirmation Elements */

= IPitomy Technical Training – Basic: Understanding the Application =

== Overview ==

 At IPitomy, we believe in crafting a PBX solution that precisely meets our customer's requirements. The key to achieving this is thorough information gathering from the customer. This information forms the basis of your IPitomy setup worksheet, a critical document defining the scope of work for system programming. Having this documented ensures alignment of expectations and serves as a reference for system reconstruction, if necessary.

= Database Building: =

==== Database Building in IPitomy's IP PBX System ====
At IPitomy, our IP PBX system represents the cutting edge of business telecommunications, functioning as a software-centric solution. This approach distinguishes it from traditional Time-Division Multiplexing (TDM) based key systems and PBXs. The primary advantage of our software-based system is its minimal reliance on hardware components, significantly reducing the likelihood of hardware failures.

Effective database construction within our system is crucial and should be meticulously planned. This planning process is vital to ensure that the final solution aligns perfectly with customer expectations. Gathering detailed information prior to the installation date is essential for a smooth, professional installation process and to minimize unbillable follow-up work.

One of the standout features of the IPitomy system is its user-friendly, web-based administration interface. This interface simplifies the programming and setup of the application. It's important to note that the IPitomy system does not include a default database. Therefore, setting up the database requires careful attention to sequence to establish the necessary elements for successful call flow.

==== Key Components of Database Building ====

# Destinations:
#* In the IPitomy system, destinations are the endpoints within the PBX where calls are routed. Examples include automated attendants or individual extensions.
#* Direct Inward Dialing (DID) numbers can be assigned to directly route calls to specific destinations, such as conference rooms or individual extensions, enhancing direct communication efficiency.
# Providers (Trunks):
#* Providers in the IPitomy system are crucial elements that manage the routing of calls both into and out of the system.
#* Once destinations are established, providers can be set up to facilitate the designed call routing requirements. This setup is integral to ensuring that calls are managed and directed according to specific business needs.

By following these steps and understanding the intricacies of our IPitomy IP PBX system, users can leverage its full potential to create a robust, efficient telecommunication setup that meets and exceeds customer expectations.
*

[[File:Menu Left detail.png|none|Menu Left detail.png]]

==== The Application Development Sequence: ====

==== Creating Extensions: ====

# ''Extension Configuration:''
#* Extensions are configurable as 3 or 4-digit numbers.
#* Features include various Call Forwarding options such as Unconditional, Busy, No Answer, or Unavailable.
#* With the creation of an extension, a corresponding voicemail box, a chat client, and a schedule are automatically generated.
#* The 'Follow-Me' feature enhances flexibility by allowing simultaneous and sequential ringing across multiple extensions or PSTN numbers.
# ''Licensing and Extension Management:''
#* The total number of available extensions is governed by the licensing agreement, with options for expansion to cater to growing business needs.
#* Separate licensing categories exist for IPitomy phones and non-IPitomy devices, with open extension licenses typically carrying a slightly higher cost.
#* The system provides Auto Provisioning for IPitomy Phones, streamlining the setup process.
#* Importing user information through a CSV file is a time-efficient method to batch-create extensions.
#* Manual creation of extensions is also an option, allowing for individualized data entry.
#* Mass editing features are available post-creation, facilitating easy management and updates of extension information.
# ''System Compatibility and Intercom Use:''
#* The IPitomy system boasts wide compatibility with a range of SIP devices, including conference phones, softphones, and other SIP-compliant equipment.
#* An added feature is the capability of using IPitomy phones for intercom paging, either between individual extensions or groups, enhancing internal communication efficiency.

'''Second'''- Create Groups of extensions.  Groups are used to ring a group of phones all at once or in a sequence. Groups are also used in paging.

*Groups are groups of extension that will ring all at once or in a round robin or other strategy
*Groups have advanced functionality for exceptionally  flexible call coverage
**Group members can be assigned a priority so individuals can be automatically assigned incoming calls during peak busy times without logging in.
**Timeout can be programmed to send calls to another destination automatically after a programmed amount of time.
**Agent ring time can be programmed to restart the ring sequence to include members who may have just wrapped up another call.
**Groups are typically used to route incoming calls so a group of phones in a department will ring.
**Groups can be used as paging zones - unicast call paging or multicast paging
**A word or digits can be pre-pended to the caller id name or number to identify which group a call is ringing in through.
**Unique Music on Hold or Message on Hold can be played per ring group

'''Third'''- Create menus and record prompts - These are automated attendants with a single digit dialing menu like - Press 1 for sales, 2 for service etc.

*Menus are single digit dialing automated attendant destinations
*Prompts can be easily recorded using the prompt recording utility in the administration interface.
*The prompts should be recorded before the menus are created
*The prompts should be planned out in advance so onsite configuration is simplified.
*Pre recorded prompts can be uploaded into the system

Once you have created the proper elements for your application, then you can create the call routing and setup and configure your trunks.  It is quick and simple if you take the time to plan out the application in advance.

'''Other Destinations'''

'''Conference''' - The conference Destination is a meet me conference room.  By default there are two conference bridge destinations; 901 and 902.  More conference bridge destinations are available by purchasing a license expansion.  The total number of members in a conference bridge is 32 people.  While you can have more than one conference call at a time, the total number of users is 32.  Greetings can be created for the conference rooms to announce what room it is; for instance Welcome to Attorney Bill Smith's private conference room.

'''Voice Mail''' - Voice mail boxes can be created.  One is automatically created for each extension that is made, so there is no need to create one for every extension, as they are already made.  Since the voice mail is integrated with the IP PBX System, there is no need to forward extensions to the voice mail; it is automatic.  The default is set to forward to voice mail after 32 seconds.

'''Schedules'''- A schedule provides the ability to route a call based upon time of day.  The schedule can route to a different destination inside of business hours, outside of business hours and during lunch.  Schedules can also be inserted at any point in the call path where time sensitive call routing is needed.

'''Branch office''' - A branch office destination creates a call route to another IPitomy IP PBX System.  Extensions in the branch office will show up and be routable using their 3 or 4 digit extension number.  A code is also available to dial to reach extensions that share the same extension number in each branch.  Calls can be routed to other branches from DID's, Menus, and Groups just like local extensions.

'''Scheduled Calls''' - Scheduled calls are a destination where an announcement or audio file can be played to a group of phones or integrated paging devices based upon time of day.  An example of this is to play a school bell audio sound to announce start and end of classes.  Emergency announcements can be made by dialing the destination extension number and entering the programmed PIN code. 

== Review and Confirmation Process in IPitomy System Setup ==

The final stage in setting up an IPitomy IP PBX system is a comprehensive review and confirmation process. This ensures complete alignment of the system components with the customer's specifications and requirements. Documenting all relevant information in the IPitomy Setup Worksheet is crucial for defining the scope of work and establishing clear expectations between the provider and the customer.

==== Extension Number Range ====

* Confirm the range of extension numbers to be used, ensuring they fit the customer's organizational structure and communication needs.

==== Device Types ====

* Document the types of devices integrated into the system, including IP phones, conference devices, and other SIP-compatible hardware.

==== Network Layout ====

* Examine and confirm the overall network layout, assessing its compatibility and efficiency for the IPitomy system.

==== Router Specifications ====

* Ensure understanding of the router's capabilities, crucial for network settings configuration and system integration.

==== Data Switch Types ====

* Identify data switch types, particularly noting Power over Ethernet (PoE) capabilities, essential for planning device connectivity and power management.

==== Cabling ====

* Check that network cabling is certified and in good condition, and that all devices and IP addresses are accurately documented.

==== Router Access ====

* Secure access to the router GUI or establish a relationship with IT personnel for necessary access.

==== IP PBX IP Address ====

* Confirm the uniqueness of the IPitomy system's IP address within the network to avoid conflicts.

==== Network Condition ====

* Assess the overall health and performance of the network, ensuring it can adequately support the IPitomy system.

==== End-User Network Responsibility ====

* Ensure that the end-user understands their role in maintaining network performance, including the necessity of network upgrades or modifications for optimal system functioning.

*

== Application Development ==

=== Voice Mail Email ===

In order for the IP PBX System to send out emails, it is necessary to have an email account assigned to the system so all emails that the system sends out can be from a legitimate email account.  This is entered into the system under <PBX Setup> <Voicemail>.  See the screen below.

Once the email settings are properly configured for sending emails, all that is required is to add the users email address in the extension.  Omitting the email address turns off the email feature in each extension.  if you are not using Voice mail to Email, do not put an email address in the email address field on the extension screen.

Email needs to be setup for sending out of the PBX to the customer's email system. 

*Sent From
***The customer must provide an Email account for the Voice Mail system (vmail@company.com) on their Email Server and use that email account so all emails from the IP PBX System will be sent using the accounts email credentials.
***If no email server is available to create an account, creating one on Google GMail (vmail.company@gmail.com) or another similar service.  Be sure to prepare for this in advance and have an email account and password ready when you go to do the installation. If you neglect to do this, it will add to your installation time.

[[File:VM to email.PNG|center|VM to email.PNG]]

=== Ring Groups/ACD ===

*Ring Groups Defined – Ring Groups are a powerful communications resource in the IPitomy IP PBX and for your customer.ACD will be discussed in the Advanced training course.<ul style="list-style-type:circle;">
</ul>

*Ring Strategy
***Go to the IPitomy WIKI at [http://wiki.ipitomy.com http://wiki.ipitomy.com]
***Select the IPitomy PBX Plus Administration Guide
***Select Groups or search for - Ring Stategy  - The WIKI will have all of the information required to set up ring groups.

**Members – those in the group with physical telephones. Members are permanent devices in the Queue. Members have the ability to place their device into Pause thereby removing themselves from queued calls, however this also eliminates all incoming calls except Page Announce Call
**Agents – those in the group with no specific telephone or wanting the ability to Log On to the Queue and Log Off the Queue separately from telephone-based functions.  Agents can log in from any phone. - Requires ACD Option
**Failovers (Queue Timeout destinations) Where the call is directed to after the timeout expires.  This can be any destiniation in the system.
**Will a menu be associated to any Ring Groups (ACD Feature – callers in an ACD - Ring Group queue can interact with options available while waiting in queue and select a new system destination)
**Use theIPitomy WIKI at [http://wiki.ipitomy.com http://wiki.ipitomy.com] to search for any of the subjects above to learn about how to implement the powerful features of Ring Groups and ACD.
**Link to Group Video Training [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

=== Menus ===

*Menus defined

*Auto Attendant, and subsequent menus must be planned in advance and well organized to allow for a streamlined installation of that portion of the application.
***Get prompts scripts – write them or have them prepared for you by the user
***Get Destination selections for one-digit dialing
***Determine if extension dialing will be allowed and at what menus
***Determine Menu overflow Destinations – where callers will be routed when they dial:
****nothing
****incorrectly
***Link to Video Training [https://www.youtube.com/watch?v=XjRzyUEpOfI&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

=== Schedules ===

*Determine schedules
*Day Hours of operation
*Lunch Hour
*Night Hours of operation
*The “Attendant” assigned telephone may be given the ability to select the Day/Night mode of operation

=== Branch Office ===

*If a Branch Office is to be part of this application the numbering plan and menus must be known for that system and it must be deployed with equal attention to detail.

Determine:

*If extension numbering at the branch office is to be transparent to the users (users may dial any extension number regardless of branch location or local PBX location with no special coding or prefix erquired). If so, the extension number scheme at the branch office must not conflict with extension numbering at this PBX or any other branch office.
*Shared Name for the Branch – this name will be applied to both PBX locations to allow them to meet (connect) over the interlinking network (Internet usually)
*Code for branch access – this code can be mirrored at each branch and used to access the paired branch
*Branch Password – this password will be applied to both PBX locations to allow them to meet (connect) over the interlinking network (Internet usually)

=== Remote Phone(s) ===

*If remote phones are to be used, assure that the router has been programmed to allow BOTH UDP and TCP Packet forwarding (Port 5060 should be configured for UDP and Ports 10,000-20,000 should be configured for UDP)
*Assure that the bandwidth at the remote phone location is adequate to handle call traffic for each telephone – especially when multiple phones are deployed at remote locations. (Plan for 200kb/s for each, two-way voice call.)
*If more than five remote phones are to be used at any remote site concurrently, consider installing an IP PBX at that location as a Branch Office instead.
*QOS is essential to all lans with VoIP traffic. We recommend setting QOS if possible to make voice call traffic a higher priority than other data traffic.
*Disable ALG (Application Layer Gateway). This router function can be powerful but a nuisance to voice traffic.
*If a Sonicwall is in use, review notes in IPitomy’s Sonicwall Configuration guide at: [http://www.ipitomy.com/webrelease/Sonicwall/Sonicwall%20Quick%20Guide.pdf http://www.ipitomy.com/webrelease/Sonicwall/Sonicwall%20Quick%20Guide.pdf]
*Go to PBX Setup=>Phone Global - enable Phone Download Enabled     - enable Phone Auth Enabled     - note the user and pass to be manually entered later (since the phone is already remote) - Click Save and Apply Changes Edit the Phone settings for the extension (pencil with handset) - Change the Configuration Updates protocol to HTTP - Click Save & Configure Phone button (were the phone local, the correct values would be sent to the phone) Manually Changing in phone - Log into phone via IP Address (user: root, pass: root) - Navigate to Phone Maintenance=>Autoprovision - Change Protocol to HTTP - Enter Username and Password from earlier Phone Global steps - Change the Software Server URL to: [http://ippbx/phonecfg/ http:///ippbx/phonecfg/]
*- Click Submit and wait for the phone to become idle
* Refer to this section of the manual : [[HD Phones#Remote Phones|Remote_Phones]]

Below is how you include the remote phones section: 

=== [[Qmanager|QManager]] ===

*Desktop Call Manager is a PC-based, Windows Application that can be loaded onto user computers to gain a high level of control of communications for their telephone. DCM is licensed per user and can be installed at on a single PBX or multiple PBX’s that are branched together with the Multisite DCM License.
*Since Desktop Call Manager integrates a Chat Client, DCM links the desktop to the world of chat and SMS Texting.
*Presence – DCM enables a presence indication via its integrated Chat client.
**DCM Provides the user with:
**Ability to monitor selected extensions on the IP PBX and Branch Office IP PBX’s
**Monitor call traffic at the monitored extensions
**Interact with call traffic at the monitored extensions
****Listen
****Whisper
****Barge
****Record calls in progress at that extension (Recordings via DCM are stored in the DCM application)

**Interact with callers in voicemail
***Screen caller leaving messages in voice mail
***Pick up (retrieve) callers from voice mail
**Record calls in progress at their own extension (Recordings via DCM are stored in the DCM application)
**Monitor Park Locations
**Monitor Trunks
**Utilize DCM-based Speed Dial
**Send and Receive Text Messages (a Chat server is required – any may be used)
**Monitor Conference Rooms 901 902… and other if licensed/programmed
**DCM may be used to invoke PBX functions:
***Dial
***Transfer
***Park
***Hang up
***Call and Extension
***Page an Extension
***Call Forward – DCM can be used to monitor PC activity and invoke pre-programmed call forward scenarios when a PC user is inactive for 15 minutes

=== Multisite Desktop Call Manager ===

*Both sites must be Multi Site Licensed and Multi Site Enabled
*MUST be enabled in the Branch office settings
*You MUST Port Forwards Ports 5048 and 5038 to the IP of the PBX at each router
*You must set up the ACL in the IP PBX to allow the mated branch office to connect on Ports 5038 and 5048. (The default only allows for local IPs).

 Questions:

#'''T'''/F, A Branch Office extension can be dialed directly without a branch office code regardless of which extension and branch where the extension is installed.
#What benefit comes from setting up a Sent From email address. '''a. puts the Sent From email address in front of the user each time an email from the IP PBX is received.''' b. puts the Sent From email address in front of the user each time an email is received. c. The email address set as Sent From becomes the server for all emails that are IP PBX generated d. The Sent From email address defines which email that are received are valid.
#T/'''F''', When DCM is used at a branch office a Multisite license must be purchased at only the IP PBX that is the Branch Office IP PBX.
#'''T'''/F, A Menu can be assigned to a ACD Group so that callers may dial a single digit and while waiting in queue and be directed to another system destination.

Where on the WWW can you go to find the IPitomy Installation and Maintenance Manual? [http://wiki.ipitomy.com http://wiki.ipitomy.com]

or

[http://www.ipitomy.com/webrelease/IPitomy/IP1100+/IPitomy%20IP1100+%20Manual.pdf http://www.ipitomy.com/webrelease/IPitomy/IP1100+/IPitomy%20IP1100+%20Manual.pdf]
[[Category:Training]]

Training:Application Solution

2023-12-11T14:42:52Z

Tyler: /* Database Building: */

= IPitomy Technical Training – Basic: Understanding the Application =

== Overview ==

 At IPitomy, we believe in crafting a PBX solution that precisely meets our customer's requirements. The key to achieving this is thorough information gathering from the customer. This information forms the basis of your IPitomy setup worksheet, a critical document defining the scope of work for system programming. Having this documented ensures alignment of expectations and serves as a reference for system reconstruction, if necessary.

= Database Building: =

==== Database Building in IPitomy's IP PBX System ====
At IPitomy, our IP PBX system represents the cutting edge of business telecommunications, functioning as a software-centric solution. This approach distinguishes it from traditional Time-Division Multiplexing (TDM) based key systems and PBXs. The primary advantage of our software-based system is its minimal reliance on hardware components, significantly reducing the likelihood of hardware failures.

Effective database construction within our system is crucial and should be meticulously planned. This planning process is vital to ensure that the final solution aligns perfectly with customer expectations. Gathering detailed information prior to the installation date is essential for a smooth, professional installation process and to minimize unbillable follow-up work.

One of the standout features of the IPitomy system is its user-friendly, web-based administration interface. This interface simplifies the programming and setup of the application. It's important to note that the IPitomy system does not include a default database. Therefore, setting up the database requires careful attention to sequence to establish the necessary elements for successful call flow.

==== Key Components of Database Building ====

# Destinations:
#* In the IPitomy system, destinations are the endpoints within the PBX where calls are routed. Examples include automated attendants or individual extensions.
#* Direct Inward Dialing (DID) numbers can be assigned to directly route calls to specific destinations, such as conference rooms or individual extensions, enhancing direct communication efficiency.
# Providers (Trunks):
#* Providers in the IPitomy system are crucial elements that manage the routing of calls both into and out of the system.
#* Once destinations are established, providers can be set up to facilitate the designed call routing requirements. This setup is integral to ensuring that calls are managed and directed according to specific business needs.

By following these steps and understanding the intricacies of our IPitomy IP PBX system, users can leverage its full potential to create a robust, efficient telecommunication setup that meets and exceeds customer expectations.
*

[[File:Menu Left detail.png|none|Menu Left detail.png]]

The Application Development Sequence:

'''First'''- create the extensions.  This can be accomplished in several ways. The module on auto provisioning covers this in detail.

About Extension:

*Extensions can be 3 or 4 digits in length
*Call Forwarding is available - Unconditional, Busy, No Answer or Unavailable.
*When an extension is created, a voicemail box, a chat client and a Schedule are created
*Follw - Me allows simultaneous and sequential ringing of any number of extension or PSTN numbers
*The quantity of Extension is subject to the license.  If additional extensions are required, they can be purchased through the license expansion.
*IPitomy Licenses are for IPitomy phones.
*Open licenses are for devices not manufactured by IPitomy.  Open extension licenses cost a little more than IPitomy extension licenses.
*Auto Provisioning is available for IPitomy Phones.
*Importing all of the users names, email addresses and extension numbers is a quick way to enter in the data and create all of the extensions at once.  This can be accomplished with a comma seperated values (.csv) file.  When the names are imported, extensions are created. or,
**Extensions can be manually created and the extension data entered one at a time
**Mass editing of the extensions saves time and allows quick changes and is possible at any time after the creation of extensions.
*IPitomy is compatible with a wide variety of SIP devices like conference phones, soft phones and anything that is SIP compatible.
*IPitomy phones can be used as intercom paging between extensions or groupd of extension.

'''Second'''- Create Groups of extensions.  Groups are used to ring a group of phones all at once or in a sequence. Groups are also used in paging.

*Groups are groups of extension that will ring all at once or in a round robin or other strategy
*Groups have advanced functionality for exceptionally  flexible call coverage
**Group members can be assigned a priority so individuals can be automatically assigned incoming calls during peak busy times without logging in.
**Timeout can be programmed to send calls to another destination automatically after a programmed amount of time.
**Agent ring time can be programmed to restart the ring sequence to include members who may have just wrapped up another call.
**Groups are typically used to route incoming calls so a group of phones in a department will ring.
**Groups can be used as paging zones - unicast call paging or multicast paging
**A word or digits can be pre-pended to the caller id name or number to identify which group a call is ringing in through.
**Unique Music on Hold or Message on Hold can be played per ring group

'''Third'''- Create menus and record prompts - These are automated attendants with a single digit dialing menu like - Press 1 for sales, 2 for service etc.

*Menus are single digit dialing automated attendant destinations
*Prompts can be easily recorded using the prompt recording utility in the administration interface.
*The prompts should be recorded before the menus are created
*The prompts should be planned out in advance so onsite configuration is simplified.
*Pre recorded prompts can be uploaded into the system

Once you have created the proper elements for your application, then you can create the call routing and setup and configure your trunks.  It is quick and simple if you take the time to plan out the application in advance.

'''Other Destinations'''

'''Conference''' - The conference Destination is a meet me conference room.  By default there are two conference bridge destinations; 901 and 902.  More conference bridge destinations are available by purchasing a license expansion.  The total number of members in a conference bridge is 32 people.  While you can have more than one conference call at a time, the total number of users is 32.  Greetings can be created for the conference rooms to announce what room it is; for instance Welcome to Attorney Bill Smith's private conference room.

'''Voice Mail''' - Voice mail boxes can be created.  One is automatically created for each extension that is made, so there is no need to create one for every extension, as they are already made.  Since the voice mail is integrated with the IP PBX System, there is no need to forward extensions to the voice mail; it is automatic.  The default is set to forward to voice mail after 32 seconds.

'''Schedules'''- A schedule provides the ability to route a call based upon time of day.  The schedule can route to a different destination inside of business hours, outside of business hours and during lunch.  Schedules can also be inserted at any point in the call path where time sensitive call routing is needed.

'''Branch office''' - A branch office destination creates a call route to another IPitomy IP PBX System.  Extensions in the branch office will show up and be routable using their 3 or 4 digit extension number.  A code is also available to dial to reach extensions that share the same extension number in each branch.  Calls can be routed to other branches from DID's, Menus, and Groups just like local extensions.

'''Scheduled Calls''' - Scheduled calls are a destination where an announcement or audio file can be played to a group of phones or integrated paging devices based upon time of day.  An example of this is to play a school bell audio sound to announce start and end of classes.  Emergency announcements can be made by dialing the destination extension number and entering the programmed PIN code. 

== Review and Confirmation Process in IPitomy System Setup ==

The final stage in setting up an IPitomy IP PBX system is a comprehensive review and confirmation process. This ensures complete alignment of the system components with the customer's specifications and requirements. Documenting all relevant information in the IPitomy Setup Worksheet is crucial for defining the scope of work and establishing clear expectations between the provider and the customer.

==== Detailed Confirmation Elements ====

==== Extension Number Range ====

* Confirm the range of extension numbers to be used, ensuring they fit the customer's organizational structure and communication needs.

==== Device Types ====

* Document the types of devices integrated into the system, including IP phones, conference devices, and other SIP-compatible hardware.
# Network Layout
#* Examine and confirm the overall network layout, assessing its compatibility and efficiency for the IPitomy system.
# Router Specifications
#* Ensure understanding of the router's capabilities, crucial for network settings configuration and system integration.
# Data Switch Types
#* Identify data switch types, particularly noting Power over Ethernet (PoE) capabilities, essential for planning device connectivity and power management.
# Cabling
#* Check that network cabling is certified and in good condition, and that all devices and IP addresses are accurately documented.
# Router Access
#* Secure access to the router GUI or establish a relationship with IT personnel for necessary access.
# IP PBX IP Address
#* Confirm the uniqueness of the IPitomy system's IP address within the network to avoid conflicts.
# Network Condition
#* Assess the overall health and performance of the network, ensuring it can adequately support the IPitomy system.
# End-User Network Responsibility
#* Ensure that the end-user understands their role in maintaining network performance, including the necessity of network upgrades or modifications for optimal system functioning.
*

== Application Development ==

=== Voice Mail Email ===

In order for the IP PBX System to send out emails, it is necessary to have an email account assigned to the system so all emails that the system sends out can be from a legitimate email account.  This is entered into the system under <PBX Setup> <Voicemail>.  See the screen below.

Once the email settings are properly configured for sending emails, all that is required is to add the users email address in the extension.  Omitting the email address turns off the email feature in each extension.  if you are not using Voice mail to Email, do not put an email address in the email address field on the extension screen.

Email needs to be setup for sending out of the PBX to the customer's email system. 

*Sent From
***The customer must provide an Email account for the Voice Mail system (vmail@company.com) on their Email Server and use that email account so all emails from the IP PBX System will be sent using the accounts email credentials.
***If no email server is available to create an account, creating one on Google GMail (vmail.company@gmail.com) or another similar service.  Be sure to prepare for this in advance and have an email account and password ready when you go to do the installation. If you neglect to do this, it will add to your installation time.

[[File:VM to email.PNG|center|VM to email.PNG]]

=== Ring Groups/ACD ===

*Ring Groups Defined – Ring Groups are a powerful communications resource in the IPitomy IP PBX and for your customer.ACD will be discussed in the Advanced training course.<ul style="list-style-type:circle;">
</ul>

*Ring Strategy
***Go to the IPitomy WIKI at [http://wiki.ipitomy.com http://wiki.ipitomy.com]
***Select the IPitomy PBX Plus Administration Guide
***Select Groups or search for - Ring Stategy  - The WIKI will have all of the information required to set up ring groups.

**Members – those in the group with physical telephones. Members are permanent devices in the Queue. Members have the ability to place their device into Pause thereby removing themselves from queued calls, however this also eliminates all incoming calls except Page Announce Call
**Agents – those in the group with no specific telephone or wanting the ability to Log On to the Queue and Log Off the Queue separately from telephone-based functions.  Agents can log in from any phone. - Requires ACD Option
**Failovers (Queue Timeout destinations) Where the call is directed to after the timeout expires.  This can be any destiniation in the system.
**Will a menu be associated to any Ring Groups (ACD Feature – callers in an ACD - Ring Group queue can interact with options available while waiting in queue and select a new system destination)
**Use theIPitomy WIKI at [http://wiki.ipitomy.com http://wiki.ipitomy.com] to search for any of the subjects above to learn about how to implement the powerful features of Ring Groups and ACD.
**Link to Group Video Training [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

=== Menus ===

*Menus defined

*Auto Attendant, and subsequent menus must be planned in advance and well organized to allow for a streamlined installation of that portion of the application.
***Get prompts scripts – write them or have them prepared for you by the user
***Get Destination selections for one-digit dialing
***Determine if extension dialing will be allowed and at what menus
***Determine Menu overflow Destinations – where callers will be routed when they dial:
****nothing
****incorrectly
***Link to Video Training [https://www.youtube.com/watch?v=XjRzyUEpOfI&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

=== Schedules ===

*Determine schedules
*Day Hours of operation
*Lunch Hour
*Night Hours of operation
*The “Attendant” assigned telephone may be given the ability to select the Day/Night mode of operation

=== Branch Office ===

*If a Branch Office is to be part of this application the numbering plan and menus must be known for that system and it must be deployed with equal attention to detail.

Determine:

*If extension numbering at the branch office is to be transparent to the users (users may dial any extension number regardless of branch location or local PBX location with no special coding or prefix erquired). If so, the extension number scheme at the branch office must not conflict with extension numbering at this PBX or any other branch office.
*Shared Name for the Branch – this name will be applied to both PBX locations to allow them to meet (connect) over the interlinking network (Internet usually)
*Code for branch access – this code can be mirrored at each branch and used to access the paired branch
*Branch Password – this password will be applied to both PBX locations to allow them to meet (connect) over the interlinking network (Internet usually)

=== Remote Phone(s) ===

*If remote phones are to be used, assure that the router has been programmed to allow BOTH UDP and TCP Packet forwarding (Port 5060 should be configured for UDP and Ports 10,000-20,000 should be configured for UDP)
*Assure that the bandwidth at the remote phone location is adequate to handle call traffic for each telephone – especially when multiple phones are deployed at remote locations. (Plan for 200kb/s for each, two-way voice call.)
*If more than five remote phones are to be used at any remote site concurrently, consider installing an IP PBX at that location as a Branch Office instead.
*QOS is essential to all lans with VoIP traffic. We recommend setting QOS if possible to make voice call traffic a higher priority than other data traffic.
*Disable ALG (Application Layer Gateway). This router function can be powerful but a nuisance to voice traffic.
*If a Sonicwall is in use, review notes in IPitomy’s Sonicwall Configuration guide at: [http://www.ipitomy.com/webrelease/Sonicwall/Sonicwall%20Quick%20Guide.pdf http://www.ipitomy.com/webrelease/Sonicwall/Sonicwall%20Quick%20Guide.pdf]
*Go to PBX Setup=>Phone Global - enable Phone Download Enabled     - enable Phone Auth Enabled     - note the user and pass to be manually entered later (since the phone is already remote) - Click Save and Apply Changes Edit the Phone settings for the extension (pencil with handset) - Change the Configuration Updates protocol to HTTP - Click Save & Configure Phone button (were the phone local, the correct values would be sent to the phone) Manually Changing in phone - Log into phone via IP Address (user: root, pass: root) - Navigate to Phone Maintenance=>Autoprovision - Change Protocol to HTTP - Enter Username and Password from earlier Phone Global steps - Change the Software Server URL to: [http://ippbx/phonecfg/ http:///ippbx/phonecfg/]
*- Click Submit and wait for the phone to become idle
* Refer to this section of the manual : [[HD Phones#Remote Phones|Remote_Phones]]

Below is how you include the remote phones section: 

=== [[Qmanager|QManager]] ===

*Desktop Call Manager is a PC-based, Windows Application that can be loaded onto user computers to gain a high level of control of communications for their telephone. DCM is licensed per user and can be installed at on a single PBX or multiple PBX’s that are branched together with the Multisite DCM License.
*Since Desktop Call Manager integrates a Chat Client, DCM links the desktop to the world of chat and SMS Texting.
*Presence – DCM enables a presence indication via its integrated Chat client.
**DCM Provides the user with:
**Ability to monitor selected extensions on the IP PBX and Branch Office IP PBX’s
**Monitor call traffic at the monitored extensions
**Interact with call traffic at the monitored extensions
****Listen
****Whisper
****Barge
****Record calls in progress at that extension (Recordings via DCM are stored in the DCM application)

**Interact with callers in voicemail
***Screen caller leaving messages in voice mail
***Pick up (retrieve) callers from voice mail
**Record calls in progress at their own extension (Recordings via DCM are stored in the DCM application)
**Monitor Park Locations
**Monitor Trunks
**Utilize DCM-based Speed Dial
**Send and Receive Text Messages (a Chat server is required – any may be used)
**Monitor Conference Rooms 901 902… and other if licensed/programmed
**DCM may be used to invoke PBX functions:
***Dial
***Transfer
***Park
***Hang up
***Call and Extension
***Page an Extension
***Call Forward – DCM can be used to monitor PC activity and invoke pre-programmed call forward scenarios when a PC user is inactive for 15 minutes

=== Multisite Desktop Call Manager ===

*Both sites must be Multi Site Licensed and Multi Site Enabled
*MUST be enabled in the Branch office settings
*You MUST Port Forwards Ports 5048 and 5038 to the IP of the PBX at each router
*You must set up the ACL in the IP PBX to allow the mated branch office to connect on Ports 5038 and 5048. (The default only allows for local IPs).

 Questions:

#'''T'''/F, A Branch Office extension can be dialed directly without a branch office code regardless of which extension and branch where the extension is installed.
#What benefit comes from setting up a Sent From email address. '''a. puts the Sent From email address in front of the user each time an email from the IP PBX is received.''' b. puts the Sent From email address in front of the user each time an email is received. c. The email address set as Sent From becomes the server for all emails that are IP PBX generated d. The Sent From email address defines which email that are received are valid.
#T/'''F''', When DCM is used at a branch office a Multisite license must be purchased at only the IP PBX that is the Branch Office IP PBX.
#'''T'''/F, A Menu can be assigned to a ACD Group so that callers may dial a single digit and while waiting in queue and be directed to another system destination.

Where on the WWW can you go to find the IPitomy Installation and Maintenance Manual? [http://wiki.ipitomy.com http://wiki.ipitomy.com]

or

[http://www.ipitomy.com/webrelease/IPitomy/IP1100+/IPitomy%20IP1100+%20Manual.pdf http://www.ipitomy.com/webrelease/IPitomy/IP1100+/IPitomy%20IP1100+%20Manual.pdf]
[[Category:Training]]

Training:Application Solution

2023-12-11T14:41:27Z

Tyler: /* Review and confirmation */

= IPitomy Technical Training – Basic: Understanding the Application =

== Overview ==

 At IPitomy, we believe in crafting a PBX solution that precisely meets our customer's requirements. The key to achieving this is thorough information gathering from the customer. This information forms the basis of your IPitomy setup worksheet, a critical document defining the scope of work for system programming. Having this documented ensures alignment of expectations and serves as a reference for system reconstruction, if necessary.

= Database Building: =

==== Database Building in IPitomy's IP PBX System ====
At IPitomy, our IP PBX system represents the cutting edge of business telecommunications, functioning as a software-centric solution. This approach distinguishes it from traditional Time-Division Multiplexing (TDM) based key systems and PBXs. The primary advantage of our software-based system is its minimal reliance on hardware components, significantly reducing the likelihood of hardware failures.

Effective database construction within our system is crucial and should be meticulously planned. This planning process is vital to ensure that the final solution aligns perfectly with customer expectations. Gathering detailed information prior to the installation date is essential for a smooth, professional installation process and to minimize unbillable follow-up work.

One of the standout features of the IPitomy system is its user-friendly, web-based administration interface. This interface simplifies the programming and setup of the application. It's important to note that the IPitomy system does not include a default database. Therefore, setting up the database requires careful attention to sequence to establish the necessary elements for successful call flow.

==== Key Components of Database Building ====

# Destinations:
#* In the IPitomy system, destinations are the endpoints within the PBX where calls are routed. Examples include automated attendants or individual extensions.
#* Direct Inward Dialing (DID) numbers can be assigned to directly route calls to specific destinations, such as conference rooms or individual extensions, enhancing direct communication efficiency.
# Providers (Trunks):
#* Providers in the IPitomy system are crucial elements that manage the routing of calls both into and out of the system.
#* Once destinations are established, providers can be set up to facilitate the designed call routing requirements. This setup is integral to ensuring that calls are managed and directed according to specific business needs.

By following these steps and understanding the intricacies of our IPitomy IP PBX system, users can leverage its full potential to create a robust, efficient telecommunication setup that meets and exceeds customer expectations.
*

[[File:Menu Left detail.png|none|Menu Left detail.png]]

The Application Development Sequence:

'''First'''- create the extensions.  This can be accomplished in several ways. The module on auto provisioning covers this in detail.

About Extension:

*Extensions can be 3 or 4 digits in length
*Call Forwarding is available - Unconditional, Busy, No Answer or Unavailable.
*When an extension is created, a voicemail box, a chat client and a Schedule are created
*Follw - Me allows simultaneous and sequential ringing of any number of extension or PSTN numbers
*The quantity of Extension is subject to the license.  If additional extensions are required, they can be purchased through the license expansion.
*IPitomy Licenses are for IPitomy phones.
*Open licenses are for devices not manufactured by IPitomy.  Open extension licenses cost a little more than IPitomy extension licenses.
*Auto Provisioning is available for IPitomy Phones.
*Importing all of the users names, email addresses and extension numbers is a quick way to enter in the data and create all of the extensions at once.  This can be accomplished with a comma seperated values (.csv) file.  When the names are imported, extensions are created. or,
**Extensions can be manually created and the extension data entered one at a time
**Mass editing of the extensions saves time and allows quick changes and is possible at any time after the creation of extensions.
*IPitomy is compatible with a wide variety of SIP devices like conference phones, soft phones and anything that is SIP compatible.
*IPitomy phones can be used as intercom paging between extensions or groupd of extension.

'''Second'''- Create Groups of extensions.  Groups are used to ring a group of phones all at once or in a sequence. Groups are also used in paging.

*Groups are groups of extension that will ring all at once or in a round robin or other strategy
*Groups have advanced functionality for exceptionally  flexible call coverage
**Group members can be assigned a priority so individuals can be automatically assigned incoming calls during peak busy times without logging in.
**Timeout can be programmed to send calls to another destination automatically after a programmed amount of time.
**Agent ring time can be programmed to restart the ring sequence to include members who may have just wrapped up another call.
**Groups are typically used to route incoming calls so a group of phones in a department will ring.
**Groups can be used as paging zones - unicast call paging or multicast paging
**A word or digits can be pre-pended to the caller id name or number to identify which group a call is ringing in through.
**Unique Music on Hold or Message on Hold can be played per ring group

'''Third'''- Create menus and record prompts - These are automated attendants with a single digit dialing menu like - Press 1 for sales, 2 for service etc.

*Menus are single digit dialing automated attendant destinations
*Prompts can be easily recorded using the prompt recording utility in the administration interface.
*The prompts should be recorded before the menus are created
*The prompts should be planned out in advance so onsite configuration is simplified.
*Pre recorded prompts can be uploaded into the system

Once you have created the proper elements for your application, then you can create the call routing and setup and configure your trunks.  It is quick and simple if you take the time to plan out the application in advance.

'''Other Destinations'''

'''Conference''' - The conference Destination is a meet me conference room.  By default there are two conference bridge destinations; 901 and 902.  More conference bridge destinations are available by purchasing a license expansion.  The total number of members in a conference bridge is 32 people.  While you can have more than one conference call at a time, the total number of users is 32.  Greetings can be created for the conference rooms to announce what room it is; for instance Welcome to Attorney Bill Smith's private conference room.

'''Voice Mail''' - Voice mail boxes can be created.  One is automatically created for each extension that is made, so there is no need to create one for every extension, as they are already made.  Since the voice mail is integrated with the IP PBX System, there is no need to forward extensions to the voice mail; it is automatic.  The default is set to forward to voice mail after 32 seconds.

'''Schedules'''- A schedule provides the ability to route a call based upon time of day.  The schedule can route to a different destination inside of business hours, outside of business hours and during lunch.  Schedules can also be inserted at any point in the call path where time sensitive call routing is needed.

'''Branch office''' - A branch office destination creates a call route to another IPitomy IP PBX System.  Extensions in the branch office will show up and be routable using their 3 or 4 digit extension number.  A code is also available to dial to reach extensions that share the same extension number in each branch.  Calls can be routed to other branches from DID's, Menus, and Groups just like local extensions.

'''Scheduled Calls''' - Scheduled calls are a destination where an announcement or audio file can be played to a group of phones or integrated paging devices based upon time of day.  An example of this is to play a school bell audio sound to announce start and end of classes.  Emergency announcements can be made by dialing the destination extension number and entering the programmed PIN code. 

== Review and Confirmation Process in IPitomy System Setup ==

The final stage in setting up an IPitomy IP PBX system is a comprehensive review and confirmation process. This ensures complete alignment of the system components with the customer's specifications and requirements. Documenting all relevant information in the IPitomy Setup Worksheet is crucial for defining the scope of work and establishing clear expectations between the provider and the customer.

==== Detailed Confirmation Elements ====

# Extension Number Range
#* Confirm the range of extension numbers to be used, ensuring they fit the customer's organizational structure and communication needs.
# Device Types
#* Document the types of devices integrated into the system, including IP phones, conference devices, and other SIP-compatible hardware.
# Network Layout
#* Examine and confirm the overall network layout, assessing its compatibility and efficiency for the IPitomy system.
# Router Specifications
#* Ensure understanding of the router's capabilities, crucial for network settings configuration and system integration.
# Data Switch Types
#* Identify data switch types, particularly noting Power over Ethernet (PoE) capabilities, essential for planning device connectivity and power management.
# Cabling
#* Check that network cabling is certified and in good condition, and that all devices and IP addresses are accurately documented.
# Router Access
#* Secure access to the router GUI or establish a relationship with IT personnel for necessary access.
# IP PBX IP Address
#* Confirm the uniqueness of the IPitomy system's IP address within the network to avoid conflicts.
# Network Condition
#* Assess the overall health and performance of the network, ensuring it can adequately support the IPitomy system.
# End-User Network Responsibility
#* Ensure that the end-user understands their role in maintaining network performance, including the necessity of network upgrades or modifications for optimal system functioning.
*

== Application Development ==

=== Voice Mail Email ===

In order for the IP PBX System to send out emails, it is necessary to have an email account assigned to the system so all emails that the system sends out can be from a legitimate email account.  This is entered into the system under <PBX Setup> <Voicemail>.  See the screen below.

Once the email settings are properly configured for sending emails, all that is required is to add the users email address in the extension.  Omitting the email address turns off the email feature in each extension.  if you are not using Voice mail to Email, do not put an email address in the email address field on the extension screen.

Email needs to be setup for sending out of the PBX to the customer's email system. 

*Sent From
***The customer must provide an Email account for the Voice Mail system (vmail@company.com) on their Email Server and use that email account so all emails from the IP PBX System will be sent using the accounts email credentials.
***If no email server is available to create an account, creating one on Google GMail (vmail.company@gmail.com) or another similar service.  Be sure to prepare for this in advance and have an email account and password ready when you go to do the installation. If you neglect to do this, it will add to your installation time.

[[File:VM to email.PNG|center|VM to email.PNG]]

=== Ring Groups/ACD ===

*Ring Groups Defined – Ring Groups are a powerful communications resource in the IPitomy IP PBX and for your customer.ACD will be discussed in the Advanced training course.<ul style="list-style-type:circle;">
</ul>

*Ring Strategy
***Go to the IPitomy WIKI at [http://wiki.ipitomy.com http://wiki.ipitomy.com]
***Select the IPitomy PBX Plus Administration Guide
***Select Groups or search for - Ring Stategy  - The WIKI will have all of the information required to set up ring groups.

**Members – those in the group with physical telephones. Members are permanent devices in the Queue. Members have the ability to place their device into Pause thereby removing themselves from queued calls, however this also eliminates all incoming calls except Page Announce Call
**Agents – those in the group with no specific telephone or wanting the ability to Log On to the Queue and Log Off the Queue separately from telephone-based functions.  Agents can log in from any phone. - Requires ACD Option
**Failovers (Queue Timeout destinations) Where the call is directed to after the timeout expires.  This can be any destiniation in the system.
**Will a menu be associated to any Ring Groups (ACD Feature – callers in an ACD - Ring Group queue can interact with options available while waiting in queue and select a new system destination)
**Use theIPitomy WIKI at [http://wiki.ipitomy.com http://wiki.ipitomy.com] to search for any of the subjects above to learn about how to implement the powerful features of Ring Groups and ACD.
**Link to Group Video Training [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

=== Menus ===

*Menus defined

*Auto Attendant, and subsequent menus must be planned in advance and well organized to allow for a streamlined installation of that portion of the application.
***Get prompts scripts – write them or have them prepared for you by the user
***Get Destination selections for one-digit dialing
***Determine if extension dialing will be allowed and at what menus
***Determine Menu overflow Destinations – where callers will be routed when they dial:
****nothing
****incorrectly
***Link to Video Training [https://www.youtube.com/watch?v=XjRzyUEpOfI&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

=== Schedules ===

*Determine schedules
*Day Hours of operation
*Lunch Hour
*Night Hours of operation
*The “Attendant” assigned telephone may be given the ability to select the Day/Night mode of operation

=== Branch Office ===

*If a Branch Office is to be part of this application the numbering plan and menus must be known for that system and it must be deployed with equal attention to detail.

Determine:

*If extension numbering at the branch office is to be transparent to the users (users may dial any extension number regardless of branch location or local PBX location with no special coding or prefix erquired). If so, the extension number scheme at the branch office must not conflict with extension numbering at this PBX or any other branch office.
*Shared Name for the Branch – this name will be applied to both PBX locations to allow them to meet (connect) over the interlinking network (Internet usually)
*Code for branch access – this code can be mirrored at each branch and used to access the paired branch
*Branch Password – this password will be applied to both PBX locations to allow them to meet (connect) over the interlinking network (Internet usually)

=== Remote Phone(s) ===

*If remote phones are to be used, assure that the router has been programmed to allow BOTH UDP and TCP Packet forwarding (Port 5060 should be configured for UDP and Ports 10,000-20,000 should be configured for UDP)
*Assure that the bandwidth at the remote phone location is adequate to handle call traffic for each telephone – especially when multiple phones are deployed at remote locations. (Plan for 200kb/s for each, two-way voice call.)
*If more than five remote phones are to be used at any remote site concurrently, consider installing an IP PBX at that location as a Branch Office instead.
*QOS is essential to all lans with VoIP traffic. We recommend setting QOS if possible to make voice call traffic a higher priority than other data traffic.
*Disable ALG (Application Layer Gateway). This router function can be powerful but a nuisance to voice traffic.
*If a Sonicwall is in use, review notes in IPitomy’s Sonicwall Configuration guide at: [http://www.ipitomy.com/webrelease/Sonicwall/Sonicwall%20Quick%20Guide.pdf http://www.ipitomy.com/webrelease/Sonicwall/Sonicwall%20Quick%20Guide.pdf]
*Go to PBX Setup=>Phone Global - enable Phone Download Enabled     - enable Phone Auth Enabled     - note the user and pass to be manually entered later (since the phone is already remote) - Click Save and Apply Changes Edit the Phone settings for the extension (pencil with handset) - Change the Configuration Updates protocol to HTTP - Click Save & Configure Phone button (were the phone local, the correct values would be sent to the phone) Manually Changing in phone - Log into phone via IP Address (user: root, pass: root) - Navigate to Phone Maintenance=>Autoprovision - Change Protocol to HTTP - Enter Username and Password from earlier Phone Global steps - Change the Software Server URL to: [http://ippbx/phonecfg/ http:///ippbx/phonecfg/]
*- Click Submit and wait for the phone to become idle
* Refer to this section of the manual : [[HD Phones#Remote Phones|Remote_Phones]]

Below is how you include the remote phones section: 

=== [[Qmanager|QManager]] ===

*Desktop Call Manager is a PC-based, Windows Application that can be loaded onto user computers to gain a high level of control of communications for their telephone. DCM is licensed per user and can be installed at on a single PBX or multiple PBX’s that are branched together with the Multisite DCM License.
*Since Desktop Call Manager integrates a Chat Client, DCM links the desktop to the world of chat and SMS Texting.
*Presence – DCM enables a presence indication via its integrated Chat client.
**DCM Provides the user with:
**Ability to monitor selected extensions on the IP PBX and Branch Office IP PBX’s
**Monitor call traffic at the monitored extensions
**Interact with call traffic at the monitored extensions
****Listen
****Whisper
****Barge
****Record calls in progress at that extension (Recordings via DCM are stored in the DCM application)

**Interact with callers in voicemail
***Screen caller leaving messages in voice mail
***Pick up (retrieve) callers from voice mail
**Record calls in progress at their own extension (Recordings via DCM are stored in the DCM application)
**Monitor Park Locations
**Monitor Trunks
**Utilize DCM-based Speed Dial
**Send and Receive Text Messages (a Chat server is required – any may be used)
**Monitor Conference Rooms 901 902… and other if licensed/programmed
**DCM may be used to invoke PBX functions:
***Dial
***Transfer
***Park
***Hang up
***Call and Extension
***Page an Extension
***Call Forward – DCM can be used to monitor PC activity and invoke pre-programmed call forward scenarios when a PC user is inactive for 15 minutes

=== Multisite Desktop Call Manager ===

*Both sites must be Multi Site Licensed and Multi Site Enabled
*MUST be enabled in the Branch office settings
*You MUST Port Forwards Ports 5048 and 5038 to the IP of the PBX at each router
*You must set up the ACL in the IP PBX to allow the mated branch office to connect on Ports 5038 and 5048. (The default only allows for local IPs).

 Questions:

#'''T'''/F, A Branch Office extension can be dialed directly without a branch office code regardless of which extension and branch where the extension is installed.
#What benefit comes from setting up a Sent From email address. '''a. puts the Sent From email address in front of the user each time an email from the IP PBX is received.''' b. puts the Sent From email address in front of the user each time an email is received. c. The email address set as Sent From becomes the server for all emails that are IP PBX generated d. The Sent From email address defines which email that are received are valid.
#T/'''F''', When DCM is used at a branch office a Multisite license must be purchased at only the IP PBX that is the Branch Office IP PBX.
#'''T'''/F, A Menu can be assigned to a ACD Group so that callers may dial a single digit and while waiting in queue and be directed to another system destination.

Where on the WWW can you go to find the IPitomy Installation and Maintenance Manual? [http://wiki.ipitomy.com http://wiki.ipitomy.com]

or

[http://www.ipitomy.com/webrelease/IPitomy/IP1100+/IPitomy%20IP1100+%20Manual.pdf http://www.ipitomy.com/webrelease/IPitomy/IP1100+/IPitomy%20IP1100+%20Manual.pdf]
[[Category:Training]]

Training:Application Solution

2023-12-11T14:39:50Z

Tyler: /* DCM (Desktop Call Manager) */

= IPitomy Technical Training – Basic: Understanding the Application =

== Overview ==

 At IPitomy, we believe in crafting a PBX solution that precisely meets our customer's requirements. The key to achieving this is thorough information gathering from the customer. This information forms the basis of your IPitomy setup worksheet, a critical document defining the scope of work for system programming. Having this documented ensures alignment of expectations and serves as a reference for system reconstruction, if necessary.

= Database Building: =

==== Database Building in IPitomy's IP PBX System ====
At IPitomy, our IP PBX system represents the cutting edge of business telecommunications, functioning as a software-centric solution. This approach distinguishes it from traditional Time-Division Multiplexing (TDM) based key systems and PBXs. The primary advantage of our software-based system is its minimal reliance on hardware components, significantly reducing the likelihood of hardware failures.

Effective database construction within our system is crucial and should be meticulously planned. This planning process is vital to ensure that the final solution aligns perfectly with customer expectations. Gathering detailed information prior to the installation date is essential for a smooth, professional installation process and to minimize unbillable follow-up work.

One of the standout features of the IPitomy system is its user-friendly, web-based administration interface. This interface simplifies the programming and setup of the application. It's important to note that the IPitomy system does not include a default database. Therefore, setting up the database requires careful attention to sequence to establish the necessary elements for successful call flow.

==== Key Components of Database Building ====

# Destinations:
#* In the IPitomy system, destinations are the endpoints within the PBX where calls are routed. Examples include automated attendants or individual extensions.
#* Direct Inward Dialing (DID) numbers can be assigned to directly route calls to specific destinations, such as conference rooms or individual extensions, enhancing direct communication efficiency.
# Providers (Trunks):
#* Providers in the IPitomy system are crucial elements that manage the routing of calls both into and out of the system.
#* Once destinations are established, providers can be set up to facilitate the designed call routing requirements. This setup is integral to ensuring that calls are managed and directed according to specific business needs.

By following these steps and understanding the intricacies of our IPitomy IP PBX system, users can leverage its full potential to create a robust, efficient telecommunication setup that meets and exceeds customer expectations.
*

[[File:Menu Left detail.png|none|Menu Left detail.png]]

The Application Development Sequence:

'''First'''- create the extensions.  This can be accomplished in several ways. The module on auto provisioning covers this in detail.

About Extension:

*Extensions can be 3 or 4 digits in length
*Call Forwarding is available - Unconditional, Busy, No Answer or Unavailable.
*When an extension is created, a voicemail box, a chat client and a Schedule are created
*Follw - Me allows simultaneous and sequential ringing of any number of extension or PSTN numbers
*The quantity of Extension is subject to the license.  If additional extensions are required, they can be purchased through the license expansion.
*IPitomy Licenses are for IPitomy phones.
*Open licenses are for devices not manufactured by IPitomy.  Open extension licenses cost a little more than IPitomy extension licenses.
*Auto Provisioning is available for IPitomy Phones.
*Importing all of the users names, email addresses and extension numbers is a quick way to enter in the data and create all of the extensions at once.  This can be accomplished with a comma seperated values (.csv) file.  When the names are imported, extensions are created. or,
**Extensions can be manually created and the extension data entered one at a time
**Mass editing of the extensions saves time and allows quick changes and is possible at any time after the creation of extensions.
*IPitomy is compatible with a wide variety of SIP devices like conference phones, soft phones and anything that is SIP compatible.
*IPitomy phones can be used as intercom paging between extensions or groupd of extension.

'''Second'''- Create Groups of extensions.  Groups are used to ring a group of phones all at once or in a sequence. Groups are also used in paging.

*Groups are groups of extension that will ring all at once or in a round robin or other strategy
*Groups have advanced functionality for exceptionally  flexible call coverage
**Group members can be assigned a priority so individuals can be automatically assigned incoming calls during peak busy times without logging in.
**Timeout can be programmed to send calls to another destination automatically after a programmed amount of time.
**Agent ring time can be programmed to restart the ring sequence to include members who may have just wrapped up another call.
**Groups are typically used to route incoming calls so a group of phones in a department will ring.
**Groups can be used as paging zones - unicast call paging or multicast paging
**A word or digits can be pre-pended to the caller id name or number to identify which group a call is ringing in through.
**Unique Music on Hold or Message on Hold can be played per ring group

'''Third'''- Create menus and record prompts - These are automated attendants with a single digit dialing menu like - Press 1 for sales, 2 for service etc.

*Menus are single digit dialing automated attendant destinations
*Prompts can be easily recorded using the prompt recording utility in the administration interface.
*The prompts should be recorded before the menus are created
*The prompts should be planned out in advance so onsite configuration is simplified.
*Pre recorded prompts can be uploaded into the system

Once you have created the proper elements for your application, then you can create the call routing and setup and configure your trunks.  It is quick and simple if you take the time to plan out the application in advance.

'''Other Destinations'''

'''Conference''' - The conference Destination is a meet me conference room.  By default there are two conference bridge destinations; 901 and 902.  More conference bridge destinations are available by purchasing a license expansion.  The total number of members in a conference bridge is 32 people.  While you can have more than one conference call at a time, the total number of users is 32.  Greetings can be created for the conference rooms to announce what room it is; for instance Welcome to Attorney Bill Smith's private conference room.

'''Voice Mail''' - Voice mail boxes can be created.  One is automatically created for each extension that is made, so there is no need to create one for every extension, as they are already made.  Since the voice mail is integrated with the IP PBX System, there is no need to forward extensions to the voice mail; it is automatic.  The default is set to forward to voice mail after 32 seconds.

'''Schedules'''- A schedule provides the ability to route a call based upon time of day.  The schedule can route to a different destination inside of business hours, outside of business hours and during lunch.  Schedules can also be inserted at any point in the call path where time sensitive call routing is needed.

'''Branch office''' - A branch office destination creates a call route to another IPitomy IP PBX System.  Extensions in the branch office will show up and be routable using their 3 or 4 digit extension number.  A code is also available to dial to reach extensions that share the same extension number in each branch.  Calls can be routed to other branches from DID's, Menus, and Groups just like local extensions.

'''Scheduled Calls''' - Scheduled calls are a destination where an announcement or audio file can be played to a group of phones or integrated paging devices based upon time of day.  An example of this is to play a school bell audio sound to announce start and end of classes.  Emergency announcements can be made by dialing the destination extension number and entering the programmed PIN code. 

== Review and confirmation ==

It is best to set up all of the end users information in the IPitomy Setup Worksheet.  This outlines your scope of work and sets the expectations between you and your customer.  Once all of the application elements have been decided, be sure to go over the worksheet with your customer.

*Extension number range
*Device Types
*Network layout
*Router type, knowledge of its capabilities – this has been covered in previous modules. The point here is KNOW it before moving on.
*Data Switch type(s), knowledge of their capabilities - POE? 
*Cabling certified and known condition – this has been covered in previous modules. All devices known and IP Addresses known
*Access to the Router GUI or relationship with IT personnel to get access when needed
*IP PBX IP Address cleared for use on the network – does not conflict with any existing device AND is guaranteed not to become in conflict with any network device
*Known condition of network
*End-user acceptance of their ownership of network performance and agreement to accept any requirement to bring their network up to “spec” should an undisclosed issue arise.

== Application Development ==

=== Voice Mail Email ===

In order for the IP PBX System to send out emails, it is necessary to have an email account assigned to the system so all emails that the system sends out can be from a legitimate email account.  This is entered into the system under <PBX Setup> <Voicemail>.  See the screen below.

Once the email settings are properly configured for sending emails, all that is required is to add the users email address in the extension.  Omitting the email address turns off the email feature in each extension.  if you are not using Voice mail to Email, do not put an email address in the email address field on the extension screen.

Email needs to be setup for sending out of the PBX to the customer's email system. 

*Sent From
***The customer must provide an Email account for the Voice Mail system (vmail@company.com) on their Email Server and use that email account so all emails from the IP PBX System will be sent using the accounts email credentials.
***If no email server is available to create an account, creating one on Google GMail (vmail.company@gmail.com) or another similar service.  Be sure to prepare for this in advance and have an email account and password ready when you go to do the installation. If you neglect to do this, it will add to your installation time.

[[File:VM to email.PNG|center|VM to email.PNG]]

=== Ring Groups/ACD ===

*Ring Groups Defined – Ring Groups are a powerful communications resource in the IPitomy IP PBX and for your customer.ACD will be discussed in the Advanced training course.<ul style="list-style-type:circle;">
</ul>

*Ring Strategy
***Go to the IPitomy WIKI at [http://wiki.ipitomy.com http://wiki.ipitomy.com]
***Select the IPitomy PBX Plus Administration Guide
***Select Groups or search for - Ring Stategy  - The WIKI will have all of the information required to set up ring groups.

**Members – those in the group with physical telephones. Members are permanent devices in the Queue. Members have the ability to place their device into Pause thereby removing themselves from queued calls, however this also eliminates all incoming calls except Page Announce Call
**Agents – those in the group with no specific telephone or wanting the ability to Log On to the Queue and Log Off the Queue separately from telephone-based functions.  Agents can log in from any phone. - Requires ACD Option
**Failovers (Queue Timeout destinations) Where the call is directed to after the timeout expires.  This can be any destiniation in the system.
**Will a menu be associated to any Ring Groups (ACD Feature – callers in an ACD - Ring Group queue can interact with options available while waiting in queue and select a new system destination)
**Use theIPitomy WIKI at [http://wiki.ipitomy.com http://wiki.ipitomy.com] to search for any of the subjects above to learn about how to implement the powerful features of Ring Groups and ACD.
**Link to Group Video Training [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

=== Menus ===

*Menus defined

*Auto Attendant, and subsequent menus must be planned in advance and well organized to allow for a streamlined installation of that portion of the application.
***Get prompts scripts – write them or have them prepared for you by the user
***Get Destination selections for one-digit dialing
***Determine if extension dialing will be allowed and at what menus
***Determine Menu overflow Destinations – where callers will be routed when they dial:
****nothing
****incorrectly
***Link to Video Training [https://www.youtube.com/watch?v=XjRzyUEpOfI&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

=== Schedules ===

*Determine schedules
*Day Hours of operation
*Lunch Hour
*Night Hours of operation
*The “Attendant” assigned telephone may be given the ability to select the Day/Night mode of operation

=== Branch Office ===

*If a Branch Office is to be part of this application the numbering plan and menus must be known for that system and it must be deployed with equal attention to detail.

Determine:

*If extension numbering at the branch office is to be transparent to the users (users may dial any extension number regardless of branch location or local PBX location with no special coding or prefix erquired). If so, the extension number scheme at the branch office must not conflict with extension numbering at this PBX or any other branch office.
*Shared Name for the Branch – this name will be applied to both PBX locations to allow them to meet (connect) over the interlinking network (Internet usually)
*Code for branch access – this code can be mirrored at each branch and used to access the paired branch
*Branch Password – this password will be applied to both PBX locations to allow them to meet (connect) over the interlinking network (Internet usually)

=== Remote Phone(s) ===

*If remote phones are to be used, assure that the router has been programmed to allow BOTH UDP and TCP Packet forwarding (Port 5060 should be configured for UDP and Ports 10,000-20,000 should be configured for UDP)
*Assure that the bandwidth at the remote phone location is adequate to handle call traffic for each telephone – especially when multiple phones are deployed at remote locations. (Plan for 200kb/s for each, two-way voice call.)
*If more than five remote phones are to be used at any remote site concurrently, consider installing an IP PBX at that location as a Branch Office instead.
*QOS is essential to all lans with VoIP traffic. We recommend setting QOS if possible to make voice call traffic a higher priority than other data traffic.
*Disable ALG (Application Layer Gateway). This router function can be powerful but a nuisance to voice traffic.
*If a Sonicwall is in use, review notes in IPitomy’s Sonicwall Configuration guide at: [http://www.ipitomy.com/webrelease/Sonicwall/Sonicwall%20Quick%20Guide.pdf http://www.ipitomy.com/webrelease/Sonicwall/Sonicwall%20Quick%20Guide.pdf]
*Go to PBX Setup=>Phone Global - enable Phone Download Enabled     - enable Phone Auth Enabled     - note the user and pass to be manually entered later (since the phone is already remote) - Click Save and Apply Changes Edit the Phone settings for the extension (pencil with handset) - Change the Configuration Updates protocol to HTTP - Click Save & Configure Phone button (were the phone local, the correct values would be sent to the phone) Manually Changing in phone - Log into phone via IP Address (user: root, pass: root) - Navigate to Phone Maintenance=>Autoprovision - Change Protocol to HTTP - Enter Username and Password from earlier Phone Global steps - Change the Software Server URL to: [http://ippbx/phonecfg/ http:///ippbx/phonecfg/]
*- Click Submit and wait for the phone to become idle
* Refer to this section of the manual : [[HD Phones#Remote Phones|Remote_Phones]]

Below is how you include the remote phones section: 

=== [[Qmanager|QManager]] ===

*Desktop Call Manager is a PC-based, Windows Application that can be loaded onto user computers to gain a high level of control of communications for their telephone. DCM is licensed per user and can be installed at on a single PBX or multiple PBX’s that are branched together with the Multisite DCM License.
*Since Desktop Call Manager integrates a Chat Client, DCM links the desktop to the world of chat and SMS Texting.
*Presence – DCM enables a presence indication via its integrated Chat client.
**DCM Provides the user with:
**Ability to monitor selected extensions on the IP PBX and Branch Office IP PBX’s
**Monitor call traffic at the monitored extensions
**Interact with call traffic at the monitored extensions
****Listen
****Whisper
****Barge
****Record calls in progress at that extension (Recordings via DCM are stored in the DCM application)

**Interact with callers in voicemail
***Screen caller leaving messages in voice mail
***Pick up (retrieve) callers from voice mail
**Record calls in progress at their own extension (Recordings via DCM are stored in the DCM application)
**Monitor Park Locations
**Monitor Trunks
**Utilize DCM-based Speed Dial
**Send and Receive Text Messages (a Chat server is required – any may be used)
**Monitor Conference Rooms 901 902… and other if licensed/programmed
**DCM may be used to invoke PBX functions:
***Dial
***Transfer
***Park
***Hang up
***Call and Extension
***Page an Extension
***Call Forward – DCM can be used to monitor PC activity and invoke pre-programmed call forward scenarios when a PC user is inactive for 15 minutes

=== Multisite Desktop Call Manager ===

*Both sites must be Multi Site Licensed and Multi Site Enabled
*MUST be enabled in the Branch office settings
*You MUST Port Forwards Ports 5048 and 5038 to the IP of the PBX at each router
*You must set up the ACL in the IP PBX to allow the mated branch office to connect on Ports 5038 and 5048. (The default only allows for local IPs).

 Questions:

#'''T'''/F, A Branch Office extension can be dialed directly without a branch office code regardless of which extension and branch where the extension is installed.
#What benefit comes from setting up a Sent From email address. '''a. puts the Sent From email address in front of the user each time an email from the IP PBX is received.''' b. puts the Sent From email address in front of the user each time an email is received. c. The email address set as Sent From becomes the server for all emails that are IP PBX generated d. The Sent From email address defines which email that are received are valid.
#T/'''F''', When DCM is used at a branch office a Multisite license must be purchased at only the IP PBX that is the Branch Office IP PBX.
#'''T'''/F, A Menu can be assigned to a ACD Group so that callers may dial a single digit and while waiting in queue and be directed to another system destination.

Where on the WWW can you go to find the IPitomy Installation and Maintenance Manual? [http://wiki.ipitomy.com http://wiki.ipitomy.com]

or

[http://www.ipitomy.com/webrelease/IPitomy/IP1100+/IPitomy%20IP1100+%20Manual.pdf http://www.ipitomy.com/webrelease/IPitomy/IP1100+/IPitomy%20IP1100+%20Manual.pdf]
[[Category:Training]]

Training: Class of Service

2023-12-11T14:04:45Z

Tyler: /* Usage examples */

== Class of Service (CoS) Training Module - IPitomy Communications ==

At IPitomy Communications, the Class of Service (CoS) feature is a fundamental aspect of our call management strategy. It allows for the customization of dialing permissions and call handling to suit specific operational needs. This training module provides comprehensive guidance on the effective utilization of CoS in various scenarios.

==== Introduction to Class of Service ====
Class of Service in IPitomy systems is a configuration of dialing permissions assigned to individual extensions. This feature enables the control of outbound routes for CoS members and can include restrictions on specific internal extensions, menus, groups, and conference rooms.

[[File:CoS edit page.JPG|File:CoS edit page.JPG]]

== Usage examples ==

==== CoS for Different Physical Locations ====
In environments with multiple locations, each may require distinct e911 settings or local caller IDs. A unique CoS can be created for each location, encompassing all standard dialing routes and a custom 911 route that employs the location's registered e911 phone number, ensuring accurate information is relayed to emergency services.

==== Managing Dialing Permissions ====
CoS is instrumental in managing specific dialing permissions, such as restricting international calls. Different CoS categories can be created based on dialing permissions, where standard routes (7, 10, 11-digit dialing) are universally available, but access to international dialing is limited to authorized users.

==== Call Recording Management ====
CoS can also be used to manage call recording settings. Various user groups can have unique CoS settings, differentiating between routes set to record calls and those that are not, addressing both compliance and privacy concerns.

==== Internal Dialing Restrictions ====
Implementing internal dialing restrictions is a key application of CoS, exemplified in setups like a lobby courtesy phone. Such a CoS configuration would allow calls to internal menus or reception groups while barring all external dialing.

==== Shared Hosted Systems ====
In shared PBX environments, CoS ensures the segregation of different organizations. Each organization's CoS restricts dialing to their specific destinations, such as extensions, groups, and conferences, maintaining privacy and security in a shared system.

==== Action Items ====

* Acquaintance with the CoS interface in the IPitomy system.
* Understanding the specific communication needs of the organization or client for effective CoS customization.
* Periodic review and update of CoS settings to reflect the changing needs of the organization.

==== Conclusion ====
Class of Service is a vital and versatile feature in IPitomy systems, offering precise control over call handling and permissions. Mastery and effective implementation of CoS are key to optimizing our communication solutions.
[[Category:Training]]

Training: Class of Service

2023-12-11T14:03:30Z

Tyler: /* Class of Service Introduction */

== Class of Service (CoS) Training Module - IPitomy Communications ==

At IPitomy Communications, the Class of Service (CoS) feature is a fundamental aspect of our call management strategy. It allows for the customization of dialing permissions and call handling to suit specific operational needs. This training module provides comprehensive guidance on the effective utilization of CoS in various scenarios.

==== Introduction to Class of Service ====
Class of Service in IPitomy systems is a configuration of dialing permissions assigned to individual extensions. This feature enables the control of outbound routes for CoS members and can include restrictions on specific internal extensions, menus, groups, and conference rooms.

[[File:CoS edit page.JPG|File:CoS edit page.JPG]]

== Usage examples ==

==== CoS for Different Physical Locations ====
In environments with multiple locations, each may require distinct e911 settings or local caller IDs. A unique CoS can be created for each location, encompassing all standard dialing routes and a custom 911 route that employs the location's registered e911 phone number, ensuring accurate information is relayed to emergency services.

==== Managing Dialing Permissions ====
CoS is instrumental in managing specific dialing permissions, such as restricting international calls. Different CoS categories can be created based on dialing permissions, where standard routes (7, 10, 11-digit dialing) are universally available, but access to international dialing is limited to authorized users.

==== Call Recording Management ====
CoS can also be used to manage call recording settings. Various user groups can have unique CoS settings, differentiating between routes set to record calls and those that are not, addressing both compliance and privacy concerns.

==== Internal Dialing Restrictions ====
Implementing internal dialing restrictions is a key application of CoS, exemplified in setups like a lobby courtesy phone. Such a CoS configuration would allow calls to internal menus or reception groups while barring all external dialing.

==== Shared Hosted Systems ====
In shared PBX environments, CoS ensures the segregation of different organizations. Each organization's CoS restricts dialing to their specific destinations, such as extensions, groups, and conferences, maintaining privacy and security in a shared system.

==== Action Items ====

* Acquaintance with the CoS interface in the IPitomy system.
* Understanding the specific communication needs of the organization or client for effective CoS customization.
* Periodic review and update of CoS settings to reflect the changing needs of the organization.

==== Conclusion ====
Class of Service is a vital and versatile feature in IPitomy systems, offering precise control over call handling and permissions. Mastery and effective implementation of CoS are key to optimizing our communication solutions.
[[Category:Training]]

Training: vLAN

2023-11-13T17:45:07Z

Tyler:

== Virtual LANs (vLANs) ==

=== Overview ===
Virtual LANs (vLANs) are a pivotal concept in modern networking, providing a logical method to segregate network traffic for reasons primarily related to security and efficiency. Unlike traditional LANs, vLANs do this without the need for separate physical networks.

=== Switch Port Types in vLANs ===

==== Access Ports ====

* Function: Designed for single vLAN membership.
* Usage: Ideal for devices like PCs or phones that don't support vLAN tagging.
* Limitation: Each port can only belong to one vLAN, limiting the use of multiple devices requiring different vLANs on the same port.

==== Trunk Ports ====

* Function: Capable of carrying multiple vLANs simultaneously.
* Configuration: Can be set to carry specific vLANs, distinguishing them using tags.
* Default Behavior: Untagged traffic is assigned to a default vLAN.

=== vLAN Traffic Types ===

==== Untagged Traffic ====

* Definition: Standard network traffic without vLAN identification.
* Assignment: Automatically assigned to the port's default vLAN.

==== Tagged Traffic ====

* Definition: Contains a vLAN ID (VID) to direct it to the appropriate vLAN.
* Security: Ports not configured for a specific VID will reject its traffic, enhancing network segmentation.

=== Setting Up vLANs ===

==== Deciding on DHCP Server ====

* Options: Choose between the router or the PBX system to manage DHCP for the vLAN.
* Considerations: This decision influences subsequent network configurations.

==== Configurations ====

===== Router as DHCP Server =====

* PBX Settings: Disable VLAN on the PBX system. Configure it to send untagged voice traffic.
* Switch Port Configuration: Set the PBX-connected port as an untagged member of the voice vLAN.

===== PBX as DHCP Server =====

* PBX Settings: Enable VLAN with a unique IP address and subnet mask. Set the maximum DHCP leases.
* Switch Port Configuration: Configure the PBX-connected port as a tagged member of the voice vLAN and untagged in the data vLAN.

=== Switch Configuration for Phones ===

* Requirement: Ports for phones should be members of both data (untagged) and voice (tagged) vLANs.
* Purpose: Ensures proper segregation of phone traffic and PC access to the data network.

=== Inter-vLAN Communication Rules ===

* Rule: Ports in a specific vLAN (e.g., vLAN 10) can only interact with ports in the same vLAN.
* Exception: Ports in different vLANs can communicate only if configured as members of the relevant vLANs.

[[File:VLAN router as DHCP.png|File:VLAN router as DHCP.png]]

== PBX Configuration for Voice vLAN ==

=== System Settings for vLAN ===

* Enable vLAN: Set to 'Enabled' to activate vLAN functionality on the PBX system.
* vLAN IP Address: Assign an IP address that the phones will use to reach the PBX. Defaulting to 10.71.66.1 is usually adequate. Ensure this is distinct from the PBX's primary network interface to avoid conflicts.
* vLAN Subnet Mask: Use 255.255.255.0 for up to 254 usable addresses, sufficient for most phone networks.
* vLAN ID (VID): Default is 10. Adjust to match the voice vLAN ID configured on your switches.
* Max DHCP Leases: Set to cover the anticipated number of phones but within the limits of your subnet (total addresses in the subnet minus one for the PBX).

=== PBX Global Phone Settings ===

* Apply vLAN Config to Phones: Set to 'Enabled' to apply vLAN settings to connected phones.
* Phone vLAN Enable: Enable this to tag voice traffic from phones.
* Phone VID: Match this with your voice vLAN ID.
* Phone Priority: Not used in current configurations.
* PC vLAN Enable: Typically disabled unless tagging PC traffic through the phone's passthrough port.
* PC VID: Set if using a data vLAN for PCs connected through phones. Requires PC vLAN to be enabled.
* PC Priority: Currently not utilized.

=== Switch Configuration for PBX and Phones ===

* PBX Port Setup: Configure the switch port connected to the PBX as:
** Untagged in the data vLAN (for communication with SIP trunks and other network services).
** Tagged in the voice vLAN (for communication with phones).
* Phone Ports Setup: Ports connecting to phones should be:
** Untagged in the data vLAN.
** Tagged in the voice vLAN, as phones will send/receive tagged voice traffic.
** PCs connected to phone passthrough ports will send their data traffic untagged.

[[Category:Training]]

Training:Verify Network

2023-11-13T17:33:47Z

Tyler: /* Optimizing the Network for Voice Traffic: A Comprehensive Guide */

== Optimizing the Network for Voice Traffic: ==

==== Introduction ====
As Voice over IP (VoIP) technology increasingly permeates enterprise and public switched voice networks, the importance of a network optimized for voice cannot be overstated. While non-voice networks may appear to function adequately, latent issues often come to light only when the more demanding requirements of voice packet transmission are introduced.

==== Understanding Network Performance for Voice ====

* Latency and Packet Transmission: Unlike data packets, voice packets are extremely sensitive to delays. A delay that is imperceptible in email transmission can be detrimental in voice communication.
* Quality of Service (QoS): Implementing QoS is critical. It prioritizes voice packets over other types of data, ensuring smooth and uninterrupted voice communication.

==== Steps to Optimize a Network for Voice ====

# Managed Switches with QoS: Employ managed switches that support Quality of Service.
#* QoS Capable Switches: Ensure switches are configured for optimal QoS.
#* Power Over Ethernet (PoE) Switches: PoE switches are recommended for VoIP applications.
#* Uninterruptible Power Supply (UPS): A UPS ensures continuity during power outages, maintaining telephone connectivity.
# Network Infrastructure: The network should be at minimum Certified CAT 5e.
#* If the current infrastructure doesn't meet this standard, prepare to upgrade the wiring and equipment.
#* Include all costs, expenses, and labor in your quote.
# Network Analysis: Conduct a thorough network analysis to identify potential issues. Consider using services like MicroConvergent for this task.
# Collaboration with IT Management: Foster a good relationship with the IT management team, especially if you are not managing the data network.

==== Testing and Certification ====

* Ensure the network is certified for voice traffic. This includes:
** Certified Category 5 cabling.
** Smart Data Switches with configured QoS.
** PoE implementation.
* For remote phone applications, ensure the router supports multiple NAT sessions. Refer to IPitomy's router compatibility guide.
* Verify adequate bandwidth for remote users and SIP Trunks. A standard guideline is 200Kb of symmetrical bandwidth for each concurrent voice call (G.711 CODEC).

==== Bandwidth Considerations ====

* Most data connections are asymmetrical. Ensure the VoIP connection aligns with the lesser bandwidth rate.
* Account for fluctuations in connection speeds throughout the day.

==== Equipment and Collaboration ====

* Always recommend appropriate equipment to avoid issues with outdated networks.
* If the data network is managed by another team, provide a checklist and a proof of assurance document for them to sign off on. This verifies that your voice network requirements are being met.

==== Checklist for Network Readiness ====

* Verify the IP address of the IP PBX.
* Establish the IP address range for DHCP use.
* Disclose any static IP addresses in use.
* Ensure switches are PoE capable and QoS enabled.
* For remote phones, confirm the router's capability for multiple NAT sessions.
* Set up necessary port forwarding to the PBX (ports 5060, 10000-20000 for remote phones, port 80 for remote maintenance, port 4569 for branch office use, and port 22 for factory support access).

==== Conclusion ====
By following the guidelines and checklist provided, you can ensure a successful IPitomy install and foster a positive, ongoing relationship with existing IT personnel. This comprehensive approach is key to integrating VoIP into your network effectively and maintaining high performance standards.
[[Category:Training]]

Training:Verify Network

2023-11-13T17:15:03Z

Tyler: /* Confirming the network is optimized for voice */

== Optimizing the Network for Voice Traffic: A Comprehensive Guide ==

==== Introduction ====
As Voice over IP (VoIP) technology increasingly permeates enterprise and public switched voice networks, the importance of a network optimized for voice cannot be overstated. While non-voice networks may appear to function adequately, latent issues often come to light only when the more demanding requirements of voice packet transmission are introduced.

==== Understanding Network Performance for Voice ====

* Latency and Packet Transmission: Unlike data packets, voice packets are extremely sensitive to delays. A delay that is imperceptible in email transmission can be detrimental in voice communication.
* Quality of Service (QoS): Implementing QoS is critical. It prioritizes voice packets over other types of data, ensuring smooth and uninterrupted voice communication.

==== Steps to Optimize a Network for Voice ====

# Managed Switches with QoS: Employ managed switches that support Quality of Service.
#* QoS Capable Switches: Ensure switches are configured for optimal QoS.
#* Power Over Ethernet (PoE) Switches: PoE switches are recommended for VoIP applications.
#* Uninterruptible Power Supply (UPS): A UPS ensures continuity during power outages, maintaining telephone connectivity.
# Network Infrastructure: The network should be at minimum Certified CAT 5e.
#* If the current infrastructure doesn't meet this standard, prepare to upgrade the wiring and equipment.
#* Include all costs, expenses, and labor in your quote.
# Network Analysis: Conduct a thorough network analysis to identify potential issues. Consider using services like MicroConvergent for this task.
# Collaboration with IT Management: Foster a good relationship with the IT management team, especially if you are not managing the data network.

==== Testing and Certification ====

* Ensure the network is certified for voice traffic. This includes:
** Certified Category 5 cabling.
** Smart Data Switches with configured QoS.
** PoE implementation.
* For remote phone applications, ensure the router supports multiple NAT sessions. Refer to IPitomy's router compatibility guide.
* Verify adequate bandwidth for remote users and SIP Trunks. A standard guideline is 200Kb of symmetrical bandwidth for each concurrent voice call (G.711 CODEC).

==== Bandwidth Considerations ====

* Most data connections are asymmetrical. Ensure the VoIP connection aligns with the lesser bandwidth rate.
* Account for fluctuations in connection speeds throughout the day.

==== Equipment and Collaboration ====

* Always recommend appropriate equipment to avoid issues with outdated networks.
* If the data network is managed by another team, provide a checklist and a proof of assurance document for them to sign off on. This verifies that your voice network requirements are being met.

==== Checklist for Network Readiness ====

* Verify the IP address of the IP PBX.
* Establish the IP address range for DHCP use.
* Disclose any static IP addresses in use.
* Ensure switches are PoE capable and QoS enabled.
* For remote phones, confirm the router's capability for multiple NAT sessions.
* Set up necessary port forwarding to the PBX (ports 5060, 10000-20000 for remote phones, port 80 for remote maintenance, port 4569 for branch office use, and port 22 for factory support access).

==== Conclusion ====
By following the guidelines and checklist provided, you can ensure a successful IPitomy install and foster a positive, ongoing relationship with existing IT personnel. This comprehensive approach is key to integrating VoIP into your network effectively and maintaining high performance standards.
[[Category:Training]]

Training:Router

2023-11-13T16:57:41Z

Tyler: /* ## **Remote Access in Network Management** */

=== Introduction to Routers ===
A router serves as the essential gateway in a network, functioning like a digital traffic controller. It manages the flow of data between your local network (LAN) and the vast expanse of the Internet. Routers perform this complex job using features like Network Address Translation (NAT), Port Forwarding, and Dynamic Host Configuration Protocol (DHCP). These functionalities ensure that your data travels efficiently, securely, and seamlessly from your devices to the web and back.

==== Network Address Translation (NAT) ====
In digital communication, IP addresses are finite. It's impractical for every device connected to the Internet to have a unique public IP address. Instead, your local network uses a private range of IP addresses, represented on the Internet by a single public IP. NAT plays a critical role here. It assigns a unique port number to each device on your network seeking Internet access. The router 'remembers' this assignment, allowing it to correctly route incoming data to the right device. Problems with NAT, like inconsistent configurations, can lead to issues such as unreachable remote phones or call reception problems.

==== Dynamic Host Configuration Protocol (DHCP) ====
Routers often double as DHCP servers, dynamically assigning IP addresses to devices on your network. In some setups, a separate server might handle DHCP. Understanding the DHCP configuration is crucial to prevent IP address conflicts. Key aspects to consider include the management of DHCP, the range of IP addresses it can assign, and the allocation of static IP addresses for critical devices like PBX systems.

 

=== Port Forwarding in Routers ===
Port forwarding is a crucial function in routers, guiding incoming internet traffic to the appropriate device within your local area network (LAN). This mechanism is especially significant in VOIP and SIP configurations. For example, remote VOIP phones often connect to your network through specific ports like 5060. To facilitate this, the router must be configured to route the traffic to the appropriate internal IP address of your PBX system.

There are three primary methods of port forwarding:

==== Single Port Forwarding ====
This method directs all incoming traffic on a specific WAN (wide area network) port to a designated LAN IP address on the same port. For example, traffic on external port 5060 can be forwarded to the PBX system at port 5060.

Table 1: Single Port Forwarding
{| class="wikitable"
!Application Name
!Port
!Protocol
!To IP Address
|-
|Remote Administration
|80
|TCP
|PBX Internal IP
|-
|SSH Support
|22
|TCP
|PBX Internal IP
|-
|SIP
|5060
|UDP
|PBX Internal IP
|-
|Branch Office
|4569
|UDP
|PBX Internal IP
|}

==== Port Range Forwarding ====
This type forwards all incoming traffic on a specified range of WAN ports to the corresponding range of ports on a LAN IP address. For instance, external ports 10000-20000 can be forwarded to the PBX system.

Table 2: Port Range Forwarding
{| class="wikitable"
!Application Name
!Port Range
!Protocol
!To IP Address
|-
|RTP
|10000-20000
|TCP & UDP
|PBX Internal IP
|}

==== 1-to-1 NAT ====
Used when a specific port is already in use. It allows for the redirection of traffic from one WAN port to a different LAN port. For example, if port 80 is used by a web server, external port 8080 can be routed to port 80 on the PBX system.

Table 3: 1 to 1 NAT
{| class="wikitable"
!Application Name
!External Port
!Internal Port
!Protocol
!To IP Address
|-
|Alternate Remote Administration
|8080
|80
|TCP
|PBX Internal IP
|}

=== Remote Access in Network Management ===
Remote access to devices such as the PBX system is a key feature for network administrators or support personnel, enabling them to modify configurations or troubleshoot issues remotely. This capability significantly enhances the efficiency of network management and technical support.

==== Configuring Remote Access to the PBX System ====
For optimal remote accessibility of the PBX system, certain ports need to be configured:

* Port 80 (Remote Admin Access): Forwarding this port allows administrators to access the PBX system's admin page from any web browser.
* Port 22 (Secure Shell or SSH Access): Forwarding this port enables secure command-line access to the PBX system.

Using this setup, administrators can access the PBX admin login page by typing <code>http://<publicIPaddress>/ippbx</code> in a web browser.

==== Dealing with Port 80 Conflicts: 1 to 1 NAT Forwarding ====
If port 80 is already in use, the 1 to 1 NAT port forwarding method should be employed. This allows mapping an alternate external port (such as 8080) to the internal port 80 of the PBX system, addressing the issue of the PBX system's fixed web access port.

Note: It's essential to maintain strong security practices, including using secure passwords and VPNs, to protect remote access points from unauthorized access. 

=== Overview of DDWRT Firmware ===
DDWRT is a well-known open-source firmware that can be installed on a wide range of router models. This firmware is celebrated for its user-friendliness and offers a standardized interface for various networking tasks, including port forwarding.

==== Configuring Port Forwarding with DDWRT ====
The port forwarding configuration interface in DDWRT is designed to be intuitive, making it accessible even for those with limited technical background. Here’s how you can set up port forwarding using the DDWRT interface:

# Accessing the Interface: After installing DDWRT firmware on your router, log in to the router's web interface. This usually involves entering the router’s IP address in a web browser.
# Navigating to Port Forwarding: In the DDWRT interface, navigate to the ‘Port Forwarding’ section. This is typically found under the ‘NAT / QoS’ menu.
# Setting Up Rules: Here, you can add port forwarding rules. This involves specifying the external port (or port range), the protocol (TCP, UDP, or both), the internal IP address to which the traffic should be directed, and the internal port if different from the external one.
# Saving and Applying Settings: After configuring the rules, save and apply the changes. The router may need to restart for the changes to take effect.

==== The Importance of Interface Familiarity ====
Understanding how to navigate and configure settings in router interfaces like DDWRT is crucial for effective network management. Whether it's setting up port forwarding, adjusting security settings, or managing DHCP, familiarity with these interfaces ensures that you can maintain a solid and secure network configuration.

==== Enhancing Network Performance and Security ====
Proper setup of features like port forwarding not only facilitates seamless communication between your network devices and the Internet but also significantly boosts the overall performance and security of your network. It's important to regularly review and update these configurations to align with changing network needs and security standards.

 

[[File:Router-PortForward.gif|none|Router-PortForward.gif]]

[[File:Router-PortRangeForwarding.gif|none|Router-PortRangeForwarding.gif]]
[[Category:Training]]

Training:Router

2023-11-13T16:55:35Z

Tyler: /* ### **Remote Access in Network Management** */

=== Introduction to Routers ===
A router serves as the essential gateway in a network, functioning like a digital traffic controller. It manages the flow of data between your local network (LAN) and the vast expanse of the Internet. Routers perform this complex job using features like Network Address Translation (NAT), Port Forwarding, and Dynamic Host Configuration Protocol (DHCP). These functionalities ensure that your data travels efficiently, securely, and seamlessly from your devices to the web and back.

==== Network Address Translation (NAT) ====
In digital communication, IP addresses are finite. It's impractical for every device connected to the Internet to have a unique public IP address. Instead, your local network uses a private range of IP addresses, represented on the Internet by a single public IP. NAT plays a critical role here. It assigns a unique port number to each device on your network seeking Internet access. The router 'remembers' this assignment, allowing it to correctly route incoming data to the right device. Problems with NAT, like inconsistent configurations, can lead to issues such as unreachable remote phones or call reception problems.

==== Dynamic Host Configuration Protocol (DHCP) ====
Routers often double as DHCP servers, dynamically assigning IP addresses to devices on your network. In some setups, a separate server might handle DHCP. Understanding the DHCP configuration is crucial to prevent IP address conflicts. Key aspects to consider include the management of DHCP, the range of IP addresses it can assign, and the allocation of static IP addresses for critical devices like PBX systems.

 

=== Port Forwarding in Routers ===
Port forwarding is a crucial function in routers, guiding incoming internet traffic to the appropriate device within your local area network (LAN). This mechanism is especially significant in VOIP and SIP configurations. For example, remote VOIP phones often connect to your network through specific ports like 5060. To facilitate this, the router must be configured to route the traffic to the appropriate internal IP address of your PBX system.

There are three primary methods of port forwarding:

==== Single Port Forwarding ====
This method directs all incoming traffic on a specific WAN (wide area network) port to a designated LAN IP address on the same port. For example, traffic on external port 5060 can be forwarded to the PBX system at port 5060.

Table 1: Single Port Forwarding
{| class="wikitable"
!Application Name
!Port
!Protocol
!To IP Address
|-
|Remote Administration
|80
|TCP
|PBX Internal IP
|-
|SSH Support
|22
|TCP
|PBX Internal IP
|-
|SIP
|5060
|UDP
|PBX Internal IP
|-
|Branch Office
|4569
|UDP
|PBX Internal IP
|}

==== Port Range Forwarding ====
This type forwards all incoming traffic on a specified range of WAN ports to the corresponding range of ports on a LAN IP address. For instance, external ports 10000-20000 can be forwarded to the PBX system.

Table 2: Port Range Forwarding
{| class="wikitable"
!Application Name
!Port Range
!Protocol
!To IP Address
|-
|RTP
|10000-20000
|TCP & UDP
|PBX Internal IP
|}

==== 1-to-1 NAT ====
Used when a specific port is already in use. It allows for the redirection of traffic from one WAN port to a different LAN port. For example, if port 80 is used by a web server, external port 8080 can be routed to port 80 on the PBX system.

Table 3: 1 to 1 NAT
{| class="wikitable"
!Application Name
!External Port
!Internal Port
!Protocol
!To IP Address
|-
|Alternate Remote Administration
|8080
|80
|TCP
|PBX Internal IP
|}
 

==== --- ====

==== ## **Remote Access in Network Management** ====

==== ### **Overview** ====

==== Remote access allows network administrators and support personnel to modify configurations or resolve issues from any location, eliminating the need for physical presence. This capability is particularly useful for managing devices like PBX systems. ====

==== ### **Optimizing PBX System Accessibility** ====

==== To enable efficient remote access: ====

==== - **Port Forwarding for Remote Management** ====

==== - **Port 80**: Used for web-based administrative access. Forwarding this port allows for remote access to the PBX system's admin panel via a web browser. ====

==== - **Port 22**: Employed for Secure Shell (SSH) access, enabling secure command-line interactions with the PBX system. ====

==== Administrators can access the PBX system remotely by entering `http://<publicIPaddress>/ippbx` in any internet-connected PC's browser. ====

==== ### **Handling Port Conflicts** ====

==== - **1-to-1 NAT for Port Conflicts**: If port 80 is already in use, 1-to-1 NAT port forwarding allows mapping an alternate external port (like 8080) to the internal port 80 of the PBX system. This step is necessary as many PBX systems do not support changing the web access port. ====

==== ### **Security Considerations** ====

==== While remote access provides convenience, it is critical to maintain security. Ensure that: ====

==== - Only authorized personnel can access these ports. ====

==== - Strong password policies are enforced. ====

==== - Network security is bolstered through the use of VPNs and regular updates of firmware and software. ====

==== --- ====

==== This formatting organizes the content into distinct sections with clear headings, making it easier to read and understand, especially for documentation or educational purposes. ====

==== Router Forwarding Interface ====
Example: DDWRT DDWRT is an open-source firmware compatible with a broad array of routers. It offers a user-friendly and relatively standardized configuration interface for setting up port forwarding. (An accompanying screenshot would showcase a router interface loaded with DDWRT Open Source firmware illustrating the configuration screen for Port Forwarding.)

Understanding and navigating these interfaces is essential for establishing and maintaining solid network configurations. Proper setup facilitates seamless communication between your network devices and the wider Internet, thereby enhancing your network's overall performance and security.

 

[[File:Router-PortForward.gif|none|Router-PortForward.gif]]

[[File:Router-PortRangeForwarding.gif|none|Router-PortRangeForwarding.gif]]
[[Category:Training]]

Training:Router

2023-11-13T16:54:11Z

Tyler: /* Introduction to Routers */

=== Introduction to Routers ===
A router serves as the essential gateway in a network, functioning like a digital traffic controller. It manages the flow of data between your local network (LAN) and the vast expanse of the Internet. Routers perform this complex job using features like Network Address Translation (NAT), Port Forwarding, and Dynamic Host Configuration Protocol (DHCP). These functionalities ensure that your data travels efficiently, securely, and seamlessly from your devices to the web and back.

==== Network Address Translation (NAT) ====
In digital communication, IP addresses are finite. It's impractical for every device connected to the Internet to have a unique public IP address. Instead, your local network uses a private range of IP addresses, represented on the Internet by a single public IP. NAT plays a critical role here. It assigns a unique port number to each device on your network seeking Internet access. The router 'remembers' this assignment, allowing it to correctly route incoming data to the right device. Problems with NAT, like inconsistent configurations, can lead to issues such as unreachable remote phones or call reception problems.

==== Dynamic Host Configuration Protocol (DHCP) ====
Routers often double as DHCP servers, dynamically assigning IP addresses to devices on your network. In some setups, a separate server might handle DHCP. Understanding the DHCP configuration is crucial to prevent IP address conflicts. Key aspects to consider include the management of DHCP, the range of IP addresses it can assign, and the allocation of static IP addresses for critical devices like PBX systems.

 

=== Port Forwarding in Routers ===
Port forwarding is a crucial function in routers, guiding incoming internet traffic to the appropriate device within your local area network (LAN). This mechanism is especially significant in VOIP and SIP configurations. For example, remote VOIP phones often connect to your network through specific ports like 5060. To facilitate this, the router must be configured to route the traffic to the appropriate internal IP address of your PBX system.

There are three primary methods of port forwarding:

==== Single Port Forwarding ====
This method directs all incoming traffic on a specific WAN (wide area network) port to a designated LAN IP address on the same port. For example, traffic on external port 5060 can be forwarded to the PBX system at port 5060.

Table 1: Single Port Forwarding
{| class="wikitable"
!Application Name
!Port
!Protocol
!To IP Address
|-
|Remote Administration
|80
|TCP
|PBX Internal IP
|-
|SSH Support
|22
|TCP
|PBX Internal IP
|-
|SIP
|5060
|UDP
|PBX Internal IP
|-
|Branch Office
|4569
|UDP
|PBX Internal IP
|}

==== Port Range Forwarding ====
This type forwards all incoming traffic on a specified range of WAN ports to the corresponding range of ports on a LAN IP address. For instance, external ports 10000-20000 can be forwarded to the PBX system.

Table 2: Port Range Forwarding
{| class="wikitable"
!Application Name
!Port Range
!Protocol
!To IP Address
|-
|RTP
|10000-20000
|TCP & UDP
|PBX Internal IP
|}

==== 1-to-1 NAT ====
Used when a specific port is already in use. It allows for the redirection of traffic from one WAN port to a different LAN port. For example, if port 80 is used by a web server, external port 8080 can be routed to port 80 on the PBX system.

Table 3: 1 to 1 NAT
{| class="wikitable"
!Application Name
!External Port
!Internal Port
!Protocol
!To IP Address
|-
|Alternate Remote Administration
|8080
|80
|TCP
|PBX Internal IP
|}
 

==== ### **Remote Access in Network Management** ====

==== Remote access to key devices like the PBX (Private Branch Exchange) system is a game-changer for network administrators and support personnel. This capability allows for the modification of configurations or troubleshooting from any location, eliminating the need for physical presence. This not only boosts the efficiency of network management but also enhances the delivery of technical support. ====

==== #### **Optimizing Accessibility of the PBX System** ====

==== For effective remote management, certain ports should be forwarded to the PBX system's internal IP address: ====

==== - **Port 80**: Commonly used for remote administrative access. Forwarding this port allows network administrators to access the PBX system's admin panel via a web browser. ====

==== - **Port 22**: Used for Secure Shell (SSH) access, enabling secure, encrypted command-line access to the system. ====

==== With the appropriate port forwarding setup, accessing the PBX system remotely is as simple as entering `http://<publicIPaddress>/ippbx` into any internet-connected PC's browser. ====

==== #### **Handling Port Conflicts with 1-to-1 NAT** ====

==== It's important to address potential port conflicts. For instance, if port 80 is already in use by another application or service in the user's network, you'll need to employ 1-to-1 NAT port forwarding. This technique allows you to map a different external port (such as 8080) to the internal port 80 of the PBX system. This is especially crucial considering that many PBX systems do not permit the alteration of the web access port. ====

==== #### **Security Considerations** ====

==== While remote access offers convenience, it's critical to balance this with security. Ensure that only authorized personnel have access to these forwarded ports. Implementing strong passwords, using VPNs (Virtual Private Networks), and regularly updating firmware and software are some ways to maintain security while benefiting from remote access capabilities. ====

==== --- ====

==== This version focuses on the importance of remote access in network management, particularly for devices like PBX systems. It includes practical advice on port forwarding and addresses potential port conflicts, all while emphasizing security considerations. ====

==== Router Forwarding Interface ====
Example: DDWRT DDWRT is an open-source firmware compatible with a broad array of routers. It offers a user-friendly and relatively standardized configuration interface for setting up port forwarding. (An accompanying screenshot would showcase a router interface loaded with DDWRT Open Source firmware illustrating the configuration screen for Port Forwarding.)

Understanding and navigating these interfaces is essential for establishing and maintaining solid network configurations. Proper setup facilitates seamless communication between your network devices and the wider Internet, thereby enhancing your network's overall performance and security.

 

[[File:Router-PortForward.gif|none|Router-PortForward.gif]]

[[File:Router-PortRangeForwarding.gif|none|Router-PortRangeForwarding.gif]]
[[Category:Training]]

IPitomy Fax

2023-08-02T17:57:59Z

Tyler: /* Installation */

__TOC__

== IPitomy E-Fax ==
IPitomy Fax Services offers a virtual fax solution for your client by Email to Fax, Web to Fax, Print to Fax, or Mobile to Fax.

[[File:How Fax Works.png|alt=|File:FaxOverview.jpg]]

== Email to Fax ==

::NOTE:: No technical fax size limit on our end, but the email provider may have a max attachment size. It is advised to keep faxes under 100 pages<gallery mode="slideshow" widths="240" heights="240">
File:EmailtoFax (1).JPG
File:EmailtoFax (2).JPG
File:EmailtoFax (3).JPG
File:EmailtoFax (4).JPG
</gallery>

Email to Fax must be sent to: destinationnumber@ipfax.net 2) Users MUST send from the email address they are registered to on the system as the system utilizes the ‘from’ email address as their Username or Login for authorization. Clients MUST be registered in the system with their email address as their Login/Username. 3) The ‘Subject’ line of the email MUST include the word ‘pass’ followed by the User’s password 4) Email to Fax is recommended be sent in Plain Text format 5) Clients may attach up to three attachments for faxing. Almost all attachment formats are supported. 6) Anything in the body of the email will be included in the cover page of the fax. An empty body will result in no cover page being sent and only the attachment(s) being faxed. 7) Clients may include the fax recipient’s name (on cover page) by including it as the first words in the ‘Subject’ field of the email. 8) Clients may include a subject for the fax by including ‘s=subject’ in the ‘Subject’ field of the email for faxing. (the word subject to be replaced by actual subject) Example of addressing of email for faxing

== Web to Fax ==
:When registering for IPitomy Fax you will receive an email with your login information. Save this email/information.
::NOTE:: Max of 3 file attachments per fax, max 2MB size per file.
<gallery widths="220" heights="220" mode="slideshow" caption="IPitomy Web to Fax">
File:Web2Fax (1).JPG
File:Web2Fax (2).JPG
File:Web2Fax (3).JPG
File:Web2Fax (4).JPG
File:Web2Fax (5).JPG
File:Web2Fax (6).JPG
File:Web2Fax (7).JPG
File:Web2Fax (8).JPG
File:Web2Fax (9).JPG
File:Web2Fax (10).JPG
File:Web2Fax (11).JPG
File:Web2Fax (12).JPG
File:Web2Fax (13).JPG
</gallery>To send a Web to Fax, a User will log into the User Portal [http://secure.ipfax.net http://secure.ipfax.net] and do the following: 1) Address who the fax is to be delivered to. (John Smith in Example) 2) Include a Fax Subject if desired. Fax Subject will be included on Cover Page of fax as well in Confirmation Report and Call Record for easy identification by the User 3) In the Fax Number(s) field, type in the fax number(s) the fax is destined for. Include country code (‘1’ for N. America but no prefix for international calls such as 011). Web to Fax can send the same fax to 10 destination numbers at the same time. 4) Upload up to three attachments. Almost all formats are supported. 5) Click Send Fax Now! . User will receive confirmation emails as configured on their account and may check Online Reports for real-time status.

== Print to Fax ==

=== Installation ===
Please Install the following program, for Windows PC's only.   MAC please use Web to Fax

IPitomy Print to Fax Installation [http://download.pangea-comm.com/ftp/printdrivers/InternetFax-v11.0.1-TLS-64bits-latest.zip Windows 64bit] [http://download.pangea-comm.com/ftp/printdrivers/InternetFax-v11.0.1-TLS-32bits-latest.zip Windows 32bit]

[[Installation Guides]]

=== Guide ===
<gallery mode="slideshow" widths="120" heights="120">
File:Print To Fax (1).JPG
File:Print To Fax (2).JPG
File:Print To Fax (3).JPG
File:Print To Fax (4).JPG
File:Print To Fax (5).JPG
File:Print To Fax (6).JPG
File:Print To Fax (7).JPG
File:Print To Fax (8).JPG
File:Print To Fax (9).JPG
File:Print To Fax (10).JPG
File:Print To Fax (11).JPG
</gallery>

IPitomy Fax

2023-08-02T17:56:31Z

Tyler: /* Installation */

__TOC__

== IPitomy E-Fax ==
IPitomy Fax Services offers a virtual fax solution for your client by Email to Fax, Web to Fax, Print to Fax, or Mobile to Fax.

[[File:How Fax Works.png|alt=|File:FaxOverview.jpg]]

== Email to Fax ==

::NOTE:: No technical fax size limit on our end, but the email provider may have a max attachment size. It is advised to keep faxes under 100 pages<gallery mode="slideshow" widths="240" heights="240">
File:EmailtoFax (1).JPG
File:EmailtoFax (2).JPG
File:EmailtoFax (3).JPG
File:EmailtoFax (4).JPG
</gallery>

Email to Fax must be sent to: destinationnumber@ipfax.net 2) Users MUST send from the email address they are registered to on the system as the system utilizes the ‘from’ email address as their Username or Login for authorization. Clients MUST be registered in the system with their email address as their Login/Username. 3) The ‘Subject’ line of the email MUST include the word ‘pass’ followed by the User’s password 4) Email to Fax is recommended be sent in Plain Text format 5) Clients may attach up to three attachments for faxing. Almost all attachment formats are supported. 6) Anything in the body of the email will be included in the cover page of the fax. An empty body will result in no cover page being sent and only the attachment(s) being faxed. 7) Clients may include the fax recipient’s name (on cover page) by including it as the first words in the ‘Subject’ field of the email. 8) Clients may include a subject for the fax by including ‘s=subject’ in the ‘Subject’ field of the email for faxing. (the word subject to be replaced by actual subject) Example of addressing of email for faxing

== Web to Fax ==
:When registering for IPitomy Fax you will receive an email with your login information. Save this email/information.
::NOTE:: Max of 3 file attachments per fax, max 2MB size per file.
<gallery widths="220" heights="220" mode="slideshow" caption="IPitomy Web to Fax">
File:Web2Fax (1).JPG
File:Web2Fax (2).JPG
File:Web2Fax (3).JPG
File:Web2Fax (4).JPG
File:Web2Fax (5).JPG
File:Web2Fax (6).JPG
File:Web2Fax (7).JPG
File:Web2Fax (8).JPG
File:Web2Fax (9).JPG
File:Web2Fax (10).JPG
File:Web2Fax (11).JPG
File:Web2Fax (12).JPG
File:Web2Fax (13).JPG
</gallery>To send a Web to Fax, a User will log into the User Portal [http://secure.ipfax.net http://secure.ipfax.net] and do the following: 1) Address who the fax is to be delivered to. (John Smith in Example) 2) Include a Fax Subject if desired. Fax Subject will be included on Cover Page of fax as well in Confirmation Report and Call Record for easy identification by the User 3) In the Fax Number(s) field, type in the fax number(s) the fax is destined for. Include country code (‘1’ for N. America but no prefix for international calls such as 011). Web to Fax can send the same fax to 10 destination numbers at the same time. 4) Upload up to three attachments. Almost all formats are supported. 5) Click Send Fax Now! . User will receive confirmation emails as configured on their account and may check Online Reports for real-time status.

== Print to Fax ==

=== Installation ===
Please Install the following program, for Windows PC's only.   MAC please use Web to Fax [http://download.pangea-comm.com/ftp/printdrivers/InternetFax-v11.0.1-TLS-64bits-latest.zip IPitomy Print to Fax Installation] - Windows 64bit [http://download.pangea-comm.com/ftp/printdrivers/InternetFax-v11.0.1-TLS-32bits-latest.zip IPitomy Print to Fax Installation] - Windows 32bit

[[Installation Guides]]

=== Guide ===
<gallery mode="slideshow" widths="120" heights="120">
File:Print To Fax (1).JPG
File:Print To Fax (2).JPG
File:Print To Fax (3).JPG
File:Print To Fax (4).JPG
File:Print To Fax (5).JPG
File:Print To Fax (6).JPG
File:Print To Fax (7).JPG
File:Print To Fax (8).JPG
File:Print To Fax (9).JPG
File:Print To Fax (10).JPG
File:Print To Fax (11).JPG
</gallery>

IPitomy Fax

2023-08-02T17:55:39Z

Tyler: /* Installation */

__TOC__

== IPitomy E-Fax ==
IPitomy Fax Services offers a virtual fax solution for your client by Email to Fax, Web to Fax, Print to Fax, or Mobile to Fax.

[[File:How Fax Works.png|alt=|File:FaxOverview.jpg]]

== Email to Fax ==

::NOTE:: No technical fax size limit on our end, but the email provider may have a max attachment size. It is advised to keep faxes under 100 pages<gallery mode="slideshow" widths="240" heights="240">
File:EmailtoFax (1).JPG
File:EmailtoFax (2).JPG
File:EmailtoFax (3).JPG
File:EmailtoFax (4).JPG
</gallery>

Email to Fax must be sent to: destinationnumber@ipfax.net 2) Users MUST send from the email address they are registered to on the system as the system utilizes the ‘from’ email address as their Username or Login for authorization. Clients MUST be registered in the system with their email address as their Login/Username. 3) The ‘Subject’ line of the email MUST include the word ‘pass’ followed by the User’s password 4) Email to Fax is recommended be sent in Plain Text format 5) Clients may attach up to three attachments for faxing. Almost all attachment formats are supported. 6) Anything in the body of the email will be included in the cover page of the fax. An empty body will result in no cover page being sent and only the attachment(s) being faxed. 7) Clients may include the fax recipient’s name (on cover page) by including it as the first words in the ‘Subject’ field of the email. 8) Clients may include a subject for the fax by including ‘s=subject’ in the ‘Subject’ field of the email for faxing. (the word subject to be replaced by actual subject) Example of addressing of email for faxing

== Web to Fax ==
:When registering for IPitomy Fax you will receive an email with your login information. Save this email/information.
::NOTE:: Max of 3 file attachments per fax, max 2MB size per file.
<gallery widths="220" heights="220" mode="slideshow" caption="IPitomy Web to Fax">
File:Web2Fax (1).JPG
File:Web2Fax (2).JPG
File:Web2Fax (3).JPG
File:Web2Fax (4).JPG
File:Web2Fax (5).JPG
File:Web2Fax (6).JPG
File:Web2Fax (7).JPG
File:Web2Fax (8).JPG
File:Web2Fax (9).JPG
File:Web2Fax (10).JPG
File:Web2Fax (11).JPG
File:Web2Fax (12).JPG
File:Web2Fax (13).JPG
</gallery>To send a Web to Fax, a User will log into the User Portal [http://secure.ipfax.net http://secure.ipfax.net] and do the following: 1) Address who the fax is to be delivered to. (John Smith in Example) 2) Include a Fax Subject if desired. Fax Subject will be included on Cover Page of fax as well in Confirmation Report and Call Record for easy identification by the User 3) In the Fax Number(s) field, type in the fax number(s) the fax is destined for. Include country code (‘1’ for N. America but no prefix for international calls such as 011). Web to Fax can send the same fax to 10 destination numbers at the same time. 4) Upload up to three attachments. Almost all formats are supported. 5) Click Send Fax Now! . User will receive confirmation emails as configured on their account and may check Online Reports for real-time status.

== Print to Fax ==

=== Installation ===
Please Install the following program, for Windows PC's only.   MAC please use Web to Fax [http://download.pangea-comm.com/ftp/printdrivers/InternetFax-v11.0.1-TLS-64bits-latest.zip IPitomy Print to Fax] [http://download.pangea-comm.com/ftp/printdrivers/InternetFax-v11.0.1-TLS-64bits-latest.zip Installation] - Windows 64bit [http://download.pangea-comm.com/ftp/printdrivers/InternetFax-v11.0.1-TLS-32bits-latest.zip IPitomy Print to Fax Installation] - Windows 32bit

[[Installation Guides]]

=== Guide ===
<gallery mode="slideshow" widths="120" heights="120">
File:Print To Fax (1).JPG
File:Print To Fax (2).JPG
File:Print To Fax (3).JPG
File:Print To Fax (4).JPG
File:Print To Fax (5).JPG
File:Print To Fax (6).JPG
File:Print To Fax (7).JPG
File:Print To Fax (8).JPG
File:Print To Fax (9).JPG
File:Print To Fax (10).JPG
File:Print To Fax (11).JPG
</gallery>

Training:Process Review

2023-07-24T16:03:49Z

Tyler: /* Remote SIP Testing */

== IPitomy PBX (On Premise) ==

==='''Process Review'''===
Preparation is critical before heading on-site to install a PBX, whether it's for an IPitomy cloud or on-premise solution. Thorough preparation reduces stress and ensures a smoother, more efficient installation process. This involves installing and testing a basic setup of the major components to ensure everything functions as expected.

=== Pre-Installation Steps ===
Thorough preparation is a key component of a successful installation. Complete the Site Survey and IPitomy Setup Worksheet, including contact information for key parties such as the ISP, IT Department, and Trunk Providers. If possible, set up as much as you can in the PBX before arriving on-site. Although it's not advisable to register the phones at this stage (since the IP address of the PBX may change), pre-configuring extensions, groups, menus, schedules, etc. reduces the time spent on-site during installation.

=== Matching the LAN ===
The first step of the installation process is configuring the PBX IP addresses to communicate on the network. If the network subnet is not 192.168.1.x, adjust the PBX to match the subnet of the LAN. There are two ways to configure the PBX IP address. After setting it up, it is recommended to reboot the PBX.

=== Using a Keyboard and Monitor ===
Connect a keyboard and monitor to the PBX and press ALT-F7. This will bring you to a screen that allows you to set the Static IP, Subnet Mask, Gateway, and DNS. Once all the values are set, select 'S' to save.

=== PC and Simple Network Setup ===
By default, you can access the PBX via 192.168.1.249/ippbx. Connect your PC and the PBX to a simple network, with only a switch between the two devices. Set your PC statically to 192.168.1.50 and log into the default IP address of the PBX. Once logged in, navigate to System => Networking and configure the Static IP, Subnet Mask, Gateway, and DNS.

=== Connecting to the Network ===
After configuring the correct IP addresses on the PBX, connect it to the customer network. Connect the PBX to the switch that will host the majority of the phones and avoid connecting it to the customer's router to prevent potential traffic bottlenecks.

=== SIP Localnet and External IP ===
Ensure the Localnet is properly configured under PBX Setup => SIP. The Localnet should match the LAN to allow phones to communicate with the PBX. The Localnet follows the pattern xxx.yyy.zzz.0, where x, y, and z match the PBX IP address, and the last octet is always zero. The subnet mask for the Localnet is typically 255.255.255.0. If remote SIP (Provider or Phones) is involved, enter the site's public IP address in the External IP field.

=== Access Control List ===
After setting the Localnet, configure the Access Control List (ACL) under System => Access Control => Access Control List. Click the 'Load Recommended Defaults' button to configure the basic ACL services (SIP, Call Manager, Local Manager, and TFTP) for devices within the Localnet to communicate with the PBX. If using a SIP provider, add <SIPTrunkIP>/32 as a rule to the SIP service in the ACL. For remote phones with static IP addresses, add them as well. If remote phones have non-static IP addresses, delete the entire SIP ACL Service and enable 'Log Watch & Ban'.

=== Registering Extensions ===
Start by registering two extensions and make test calls to ensure everything is functioning correctly. Verify if each phone can call the other, if there is two-way audio, and if there are any issues with call quality.

=== Testing Remote SIP ===
If the site plans to use remote phones or SIP trunks, install a remote phone to test if the router is handling NAT correctly. It's recommended to identify any issues at the beginning of the installation to allow time for router adjustments if necessary.

=== Testing with Softphones and Hardware Phones ===
Use a SIP softphone on a cell phone to test WAN extension registration to the PBX. Also, have other employees register physical SIP phones to WAN extensions on the PBX and test.

=== Configuring Trunks ===
Configure and test trunks early in the installation process. This allows the provider time to resolve any possible issues while you work on the rest of the installation. Add only one DID at this time to ensure the provider is sending the correct number of digits.

=== Thorough Testing ===
Thoroughly test the installation by setting up local extensions, remote extensions, and trunks. Ensure LAN phones can make and receive calls, DTMF works correctly, remote phones can make and receive calls, and trunks function properly. Verify there is two-way audio for LAN phones, WAN phones, and trunks, and check if DIDs are routing correctly.

=== Complete Configuration ===
After the basic installation has been tested and is functioning correctly, register the remaining phones to the PBX, add and configure the remaining DIDs, and thoroughly test the complete functionality. Check if Ring Group calls function as desired and if Menus route callers to the intended destinations.

=== Training ===
Once the system is installed and functioning as expected, begin training the end users. Many features will work similarly to their old system, but there may be new things to learn. Ensuring that end users are familiar with their phones and the PBX will result in satisfied customers.

== IPitomy Cloud PBX ==
IPitomy Cloud PBX Process Review Proper preparation is essential before initiating the setup of IPitomy's Cloud PBX solution. This includes creating a blueprint of the entire network setup, including SIP Firewall & Routers, user extensions, and phone types which ensures an efficient and smooth installation process.

=== Pre-Installation Steps ===
Effective preparation is crucial to the successful implementation of a Cloud PBX solution. Completing the IPitomy Setup Worksheet, which includes important contact details for the ISP, IT Department, and Trunk Providers, is an essential first step. Pre-configuring extensions, groups, menus, schedules, etc., as much as possible before the implementation starts can significantly reduce the time required during the actual setup.

=== Cloud Configuration ===
Once the pre-installation steps are complete, configure the Cloud PBX to match the necessary specifications and requirements. This step includes setting up and disabling the appropriate firewall settings to ensure secure SIP communications, configuring necessary port forwarding rules if required, and preparing the system for integration with the network. If an ACL (Access Control List) is being implemented make sure to whitelist the approved IP addresses on your IPitomy Cloud PBX.

===Access Control List===
Only enable an ACL Policy when static IP addresses are in use! Configure the Access Control List (ACL) under System => Access Control => Access Control List. New Services may use an allow or deny list, giving you the ability to control what IP addresses can and or cannot communicate with your system.

=== Web Interface Access ===
Access the IPitomy Cloud PBX interface via the provided URL. Once logged in, navigate through the system settings to configure the network settings, SIP trunk parameters, user extensions, and other necessary details as per the initial network blueprint.

=== Extension Setup ===
Set up the required extensions for users on the Cloud PBX system. This includes assigning extension numbers, setting up voicemail, configuring call forwarding rules if necessary, and providing the required credentials to the users for setting up their phones or softphones.

=== Remote SIP Testing ===
If the setup includes additional remote phones test them to ensure their router is handling Network Address Translation (NAT) correctly. It's essential to detect any potential issues early in the installation to allow time for necessary adjustments. If you are deploying remote phones at multiple locations you should always test and add an additional E911 end point for each location.

=== Softphone and Hardware Phone Testing ===
Use a SIP softphone on a mobile device to test the registration of extensions to the Cloud PBX. Also, when deploying remote phones have employees register their physical or soft SIP phones to extensions on the Cloud PBX and conduct tests to ensure they work properly.

=== Comprehensive Testing ===
Thoroughly test the system by setting up local extensions, remote extensions, and trunks. Confirm that all phones can make and receive calls, that DTMF is functioning correctly, and that all trunks are operating as they should. Check that there is two-way audio on all phones and that DIDs are routing correctly.

=== Final Configuration ===
After all tests confirm that the basic setup is functioning correctly, finalize the Cloud PBX configuration. This includes adding and configuring any remaining DIDs, setting up any additional features like call queues or IVRs, and testing the complete system to ensure everything is working as expected.

=== User Training ===
Once the Cloud PBX system is fully set up and functional, begin training the end users. While many features may be similar to their previous system, there will likely be new features and functions to learn. Ensuring that end users are comfortable with their phones and the new PBX system will result in satisfied customers.

[[Category:Training]]

Training:Process Review

2023-07-13T13:36:13Z

Tyler: /* IPitomy Cloud PBX */

== IPitomy PBX (On Premise) ==

==='''Process Review'''===
Preparation is critical before heading on-site to install a PBX, whether it's for an IPitomy cloud or on-premise solution. Thorough preparation reduces stress and ensures a smoother, more efficient installation process. This involves installing and testing a basic setup of the major components to ensure everything functions as expected.

=== Pre-Installation Steps ===
Thorough preparation is a key component of a successful installation. Complete the Site Survey and IPitomy Setup Worksheet, including contact information for key parties such as the ISP, IT Department, and Trunk Providers. If possible, set up as much as you can in the PBX before arriving on-site. Although it's not advisable to register the phones at this stage (since the IP address of the PBX may change), pre-configuring extensions, groups, menus, schedules, etc. reduces the time spent on-site during installation.

=== Matching the LAN ===
The first step of the installation process is configuring the PBX IP addresses to communicate on the network. If the network subnet is not 192.168.1.x, adjust the PBX to match the subnet of the LAN. There are two ways to configure the PBX IP address. After setting it up, it is recommended to reboot the PBX.

=== Using a Keyboard and Monitor ===
Connect a keyboard and monitor to the PBX and press ALT-F7. This will bring you to a screen that allows you to set the Static IP, Subnet Mask, Gateway, and DNS. Once all the values are set, select 'S' to save.

=== PC and Simple Network Setup ===
By default, you can access the PBX via 192.168.1.249/ippbx. Connect your PC and the PBX to a simple network, with only a switch between the two devices. Set your PC statically to 192.168.1.50 and log into the default IP address of the PBX. Once logged in, navigate to System => Networking and configure the Static IP, Subnet Mask, Gateway, and DNS.

=== Connecting to the Network ===
After configuring the correct IP addresses on the PBX, connect it to the customer network. Connect the PBX to the switch that will host the majority of the phones and avoid connecting it to the customer's router to prevent potential traffic bottlenecks.

=== SIP Localnet and External IP ===
Ensure the Localnet is properly configured under PBX Setup => SIP. The Localnet should match the LAN to allow phones to communicate with the PBX. The Localnet follows the pattern xxx.yyy.zzz.0, where x, y, and z match the PBX IP address, and the last octet is always zero. The subnet mask for the Localnet is typically 255.255.255.0. If remote SIP (Provider or Phones) is involved, enter the site's public IP address in the External IP field.

=== Access Control List ===
After setting the Localnet, configure the Access Control List (ACL) under System => Access Control => Access Control List. Click the 'Load Recommended Defaults' button to configure the basic ACL services (SIP, Call Manager, Local Manager, and TFTP) for devices within the Localnet to communicate with the PBX. If using a SIP provider, add <SIPTrunkIP>/32 as a rule to the SIP service in the ACL. For remote phones with static IP addresses, add them as well. If remote phones have non-static IP addresses, delete the entire SIP ACL Service and enable 'Log Watch & Ban'.

=== Registering Extensions ===
Start by registering two extensions and make test calls to ensure everything is functioning correctly. Verify if each phone can call the other, if there is two-way audio, and if there are any issues with call quality.

=== Testing Remote SIP ===
If the site plans to use remote phones or SIP trunks, install a remote phone to test if the router is handling NAT correctly. It's recommended to identify any issues at the beginning of the installation to allow time for router adjustments if necessary.

=== Testing with Softphones and Hardware Phones ===
Use a SIP softphone on a cell phone to test WAN extension registration to the PBX. Also, have other employees register physical SIP phones to WAN extensions on the PBX and test.

=== Configuring Trunks ===
Configure and test trunks early in the installation process. This allows the provider time to resolve any possible issues while you work on the rest of the installation. Add only one DID at this time to ensure the provider is sending the correct number of digits.

=== Thorough Testing ===
Thoroughly test the installation by setting up local extensions, remote extensions, and trunks. Ensure LAN phones can make and receive calls, DTMF works correctly, remote phones can make and receive calls, and trunks function properly. Verify there is two-way audio for LAN phones, WAN phones, and trunks, and check if DIDs are routing correctly.

=== Complete Configuration ===
After the basic installation has been tested and is functioning correctly, register the remaining phones to the PBX, add and configure the remaining DIDs, and thoroughly test the complete functionality. Check if Ring Group calls function as desired and if Menus route callers to the intended destinations.

=== Training ===
Once the system is installed and functioning as expected, begin training the end users. Many features will work similarly to their old system, but there may be new things to learn. Ensuring that end users are familiar with their phones and the PBX will result in satisfied customers.

== IPitomy Cloud PBX ==
IPitomy Cloud PBX Process Review Proper preparation is essential before initiating the setup of IPitomy's Cloud PBX solution. This includes creating a blueprint of the entire network setup, including SIP Firewall & Routers, user extensions, and phone types which ensures an efficient and smooth installation process.

=== Pre-Installation Steps ===
Effective preparation is crucial to the successful implementation of a Cloud PBX solution. Completing the IPitomy Setup Worksheet, which includes important contact details for the ISP, IT Department, and Trunk Providers, is an essential first step. Pre-configuring extensions, groups, menus, schedules, etc., as much as possible before the implementation starts can significantly reduce the time required during the actual setup.

=== Cloud Configuration ===
Once the pre-installation steps are complete, configure the Cloud PBX to match the necessary specifications and requirements. This step includes setting up and disabling the appropriate firewall settings to ensure secure SIP communications, configuring necessary port forwarding rules if required, and preparing the system for integration with the network.

=== Web Interface Access ===
Access the IPitomy Cloud PBX interface via the provided URL. Once logged in, navigate through the system settings to configure the network settings, SIP trunk parameters, user extensions, and other necessary details as per the initial network blueprint.

=== Extension Setup ===
Set up the required extensions for users on the Cloud PBX system. This includes assigning extension numbers, setting up voicemail, configuring call forwarding rules if necessary, and providing the required credentials to the users for setting up their phones or softphones.

=== Remote SIP Testing ===
If the setup includes remote phones or SIP trunks, test them to ensure the router is handling Network Address Translation (NAT) correctly. It's essential to detect any potential issues early in the installation to allow time for necessary adjustments. If you are deploying remote phones at multiple locations you should always add an additional E911 end point for each location.

=== Softphone and Hardware Phone Testing ===
Use a SIP softphone on a mobile device to test the registration of WAN extensions to the Cloud PBX. Also, when deploying remote phones have employees register their physical or soft SIP phones to extensions on the Cloud PBX and conduct tests to ensure they work properly.

=== Comprehensive Testing ===
Thoroughly test the system by setting up local extensions, remote extensions, and trunks. Confirm that all phones can make and receive calls, that DTMF is functioning correctly, and that all trunks are operating as they should. Check that there is two-way audio on all phones and that DIDs are routing correctly.

=== Final Configuration ===
After all tests confirm that the basic setup is functioning correctly, finalize the Cloud PBX configuration. This includes adding and configuring any remaining DIDs, setting up any additional features like call queues or IVRs, and testing the complete system to ensure everything is working as expected.

=== User Training ===
Once the Cloud PBX system is fully set up and functional, begin training the end users. While many features may be similar to their previous system, there will likely be new features and functions to learn. Ensuring that end users are comfortable with their phones and the new PBX system will result in satisfied customers.

[[Category:Training]]

Training:Process Review

2023-07-13T13:33:35Z

Tyler: /* IPitomy Cloud PBX */

== IPitomy PBX (On Premise) ==

==== '''Process Review''' ====
Preparation is critical before heading on-site to install a PBX, whether it's for an IPitomy cloud or on-premise solution. Thorough preparation reduces stress and ensures a smoother, more efficient installation process. This involves installing and testing a basic setup of the major components to ensure everything functions as expected.

==== Pre-Installation Steps ====
Thorough preparation is a key component of a successful installation. Complete the Site Survey and IPitomy Setup Worksheet, including contact information for key parties such as the ISP, IT Department, and Trunk Providers. If possible, set up as much as you can in the PBX before arriving on-site. Although it's not advisable to register the phones at this stage (since the IP address of the PBX may change), pre-configuring extensions, groups, menus, schedules, etc. reduces the time spent on-site during installation.

==== Matching the LAN ====
The first step of the installation process is configuring the PBX IP addresses to communicate on the network. If the network subnet is not 192.168.1.x, adjust the PBX to match the subnet of the LAN. There are two ways to configure the PBX IP address. After setting it up, it is recommended to reboot the PBX.

==== Using a Keyboard and Monitor ====
Connect a keyboard and monitor to the PBX and press ALT-F7. This will bring you to a screen that allows you to set the Static IP, Subnet Mask, Gateway, and DNS. Once all the values are set, select 'S' to save.

==== PC and Simple Network Setup ====
By default, you can access the PBX via 192.168.1.249/ippbx. Connect your PC and the PBX to a simple network, with only a switch between the two devices. Set your PC statically to 192.168.1.50 and log into the default IP address of the PBX. Once logged in, navigate to System => Networking and configure the Static IP, Subnet Mask, Gateway, and DNS.

==== Connecting to the Network ====
After configuring the correct IP addresses on the PBX, connect it to the customer network. Connect the PBX to the switch that will host the majority of the phones and avoid connecting it to the customer's router to prevent potential traffic bottlenecks.

==== SIP Localnet and External IP ====
Ensure the Localnet is properly configured under PBX Setup => SIP. The Localnet should match the LAN to allow phones to communicate with the PBX. The Localnet follows the pattern xxx.yyy.zzz.0, where x, y, and z match the PBX IP address, and the last octet is always zero. The subnet mask for the Localnet is typically 255.255.255.0. If remote SIP (Provider or Phones) is involved, enter the site's public IP address in the External IP field.

==== Access Control List ====
After setting the Localnet, configure the Access Control List (ACL) under System => Access Control => Access Control List. Click the 'Load Recommended Defaults' button to configure the basic ACL services (SIP, Call Manager, Local Manager, and TFTP) for devices within the Localnet to communicate with the PBX. If using a SIP provider, add <SIPTrunkIP>/32 as a rule to the SIP service in the ACL. For remote phones with static IP addresses, add them as well. If remote phones have non-static IP addresses, delete the entire SIP ACL Service and enable 'Log Watch & Ban'.

==== Registering Extensions ====
Start by registering two extensions and make test calls to ensure everything is functioning correctly. Verify if each phone can call the other, if there is two-way audio, and if there are any issues with call quality.

==== Testing Remote SIP ====
If the site plans to use remote phones or SIP trunks, install a remote phone to test if the router is handling NAT correctly. It's recommended to identify any issues at the beginning of the installation to allow time for router adjustments if necessary.

==== Testing with Softphones and Hardware Phones ====
Use a SIP softphone on a cell phone to test WAN extension registration to the PBX. Also, have other employees register physical SIP phones to WAN extensions on the PBX and test.

==== Configuring Trunks ====
Configure and test trunks early in the installation process. This allows the provider time to resolve any possible issues while you work on the rest of the installation. Add only one DID at this time to ensure the provider is sending the correct number of digits.

==== Thorough Testing ====
Thoroughly test the installation by setting up local extensions, remote extensions, and trunks. Ensure LAN phones can make and receive calls, DTMF works correctly, remote phones can make and receive calls, and trunks function properly. Verify there is two-way audio for LAN phones, WAN phones, and trunks, and check if DIDs are routing correctly.

==== Complete Configuration ====
After the basic installation has been tested and is functioning correctly, register the remaining phones to the PBX, add and configure the remaining DIDs, and thoroughly test the complete functionality. Check if Ring Group calls function as desired and if Menus route callers to the intended destinations.

==== Training ====
Once the system is installed and functioning as expected, begin training the end users. Many features will work similarly to their old system, but there may be new things to learn. Ensuring that end users are familiar with their phones and the PBX will result in satisfied customers.

== IPitomy Cloud PBX ==
IPitomy Cloud PBX Process Review Proper preparation is essential before initiating the setup of IPitomy's Cloud PBX solution. This includes creating a blueprint of the entire network setup, including SIP Firewall & Routers, user extensions, and phone types which ensures an efficient and smooth installation process.

=== Pre-Installation Steps ===
Effective preparation is crucial to the successful implementation of a Cloud PBX solution. Completing the IPitomy Setup Worksheet, which includes important contact details for the ISP, IT Department, and Trunk Providers, is an essential first step. Pre-configuring extensions, groups, menus, schedules, etc., as much as possible before the implementation starts can significantly reduce the time required during the actual setup.

=== Cloud Configuration ===
Once the pre-installation steps are complete, configure the Cloud PBX to match the necessary specifications and requirements. This step includes setting up and disabling the appropriate firewall settings to ensure secure SIP communications, configuring necessary port forwarding rules if required, and preparing the system for integration with the network.

=== Web Interface Access ===
Access the IPitomy Cloud PBX interface via the provided URL. Once logged in, navigate through the system settings to configure the network settings, SIP trunk parameters, user extensions, and other necessary details as per the initial network blueprint.

=== Extension Setup ===
Set up the required extensions for users on the Cloud PBX system. This includes assigning extension numbers, setting up voicemail, configuring call forwarding rules if necessary, and providing the required credentials to the users for setting up their phones or softphones.

=== Remote SIP Testing ===
If the setup includes remote phones or SIP trunks, test them to ensure the router is handling Network Address Translation (NAT) correctly. It's essential to detect any potential issues early in the installation to allow time for necessary adjustments. If you are deploying remote phones at multiple locations you should always add an additional E911 end point for each location.

=== Softphone and Hardware Phone Testing ===
Use a SIP softphone on a mobile device to test the registration of WAN extensions to the Cloud PBX. Also, when deploying remote phones have employees register their physical or soft SIP phones to extensions on the Cloud PBX and conduct tests to ensure they work properly.

=== Comprehensive Testing ===
Thoroughly test the system by setting up local extensions, remote extensions, and trunks. Confirm that all phones can make and receive calls, that DTMF is functioning correctly, and that all trunks are operating as they should. Check that there is two-way audio on all phones and that DIDs are routing correctly.

=== Final Configuration ===
After all tests confirm that the basic setup is functioning correctly, finalize the Cloud PBX configuration. This includes adding and configuring any remaining DIDs, setting up any additional features like call queues or IVRs, and testing the complete system to ensure everything is working as expected.

=== User Training ===
Once the Cloud PBX system is fully set up and functional, begin training the end users. While many features may be similar to their previous system, there will likely be new features and functions to learn. Ensuring that end users are comfortable with their phones and the new PBX system will result in satisfied customers.

[[Category:Training]]

Training:Router

2023-06-20T18:58:27Z

Tyler:

=== Introduction to Routers ===
A router is a pivotal component in any network structure, functioning as the digital 'traffic controller,' managing data flow between your local area network (LAN) and the broader Internet. The router accomplishes this intricate task through features such as Network Address Translation (NAT), Port Forwarding, and Dynamic Host Configuration Protocol (DHCP), facilitating seamless, secure, and efficient data transmission from your devices to the web and vice versa.

==== Network Address Translation (NAT) ====
In the realm of digital communications, IP addresses are limited. Consequently, not every device linked to the Internet can possess a unique public IP address. Rather, your local network maintains a distinct private subnet of IP addresses, represented on the Internet by a single public IP. This is where NAT comes into play. Whenever a device on your network seeks Internet connectivity, the router uses NAT to associate a unique port number with that device. The router then 'remembers' this association, enabling it to direct responses accurately. NAT-related issues, such as inconsistent NAT, often result in connectivity problems. For example, inconsistent NAT might render remote phones unreachable or incapable of receiving calls.

==== Dynamic Host Configuration Protocol (DHCP) ====
Routers typically act as DHCP servers, assigning IP addresses to network devices. However, in certain scenarios, a standalone server within the network might undertake the DHCP function. Recognizing the DHCP setup during a site survey is vital to avoid IP conflicts. You need to ascertain how DHCP will be managed, the DHCP range, and an inventory of available static IP addresses for configuring devices like PBX systems.

==== Port Forwarding ====
Port forwarding is a router's method of ensuring that incoming packets on specific ports are directed to the appropriate device within the LAN. For instance, remote phones initiate communication by dispatching packets to port 5060. Therefore, this port needs to be forwarded in the router to the internal static IP address of the PBX.

Port forwarding can be categorized into three types:

Single Port Forwarding: All incoming WAN traffic on a specific port is directed to a certain LAN IP via that port. For instance, external port 5060 can be forwarded to the PBX IP on port 5060.

Port Range Forwarding: All incoming WAN traffic on a range of ports is directed to a certain LAN IP via that range of ports. For instance, external ports 10000 to 20000 can be forwarded to the PBX IP on ports 10000 to 20000.

1 to 1 NAT: All incoming WAN traffic on a specific port is directed to a certain LAN IP via a different port. This is usually utilized when the required port is already occupied. For example, if a user hosts their own webpage and port 80 is used, you can forward external port 8080 to the PBX IP on port 80.

A correct router configuration is critical to maintaining a stable and secure network. Misconfigurations can lead to inaccessible devices, security risks, or even total network shutdown. Therefore, mastering the management of these fundamental router functions is a crucial skill for network professionals.

 

Table 1: Single Port Forwarding

{| border="1" cellpadding="0" cellspacing="0" width="345"
|-
| Application Name
| Port
| Protocol
| To IP Address
|-
| Remote Administration
| 80
| TCP
| PBX Internal IP
|-
| SSH Support
| 22
| TCP
| PBX Internal IP
|-
| SIP
| 5060
| UDP
| PBX Internal IP
|-
| Branch Office
| 4569
| UDP
| PBX Internal IP
|}

 

Table 2: Port Range Forwarding

{| border="1" cellpadding="0" cellspacing="0" width="364"
|-
| Application Name
| Port
| Protocol
| To IP Address
|-
| RTP
| 10000-20000
| TCP & UDP
| PBX Internal IP
|}

 

Table 3: 1 to 1 NAT

{| border="1" cellpadding="0" cellspacing="0" height="94" width="523"
|-
| Application Name
| External Port
| Internal Port
| Protocol
| To IP Address
|-
| Alternate Remote Administration
| 8080
| 80
| TCP
| PBX Internal IP
|}

 

 

==== Remote access ====
to devices such as the PBX system empowers network administrators or support personnel to modify configurations or resolve issues from any location, bypassing the need for on-site presence. This functionality amplifies the efficiency of network management and technical support provision. For optimal accessibility and visibility of the PBX system, it's advised to forward port 80 (utilized for remote admin access) and port 22 (utilized for Secure Shell or SSH access) to the PBX's internal IP address. With this setup, you can input <publicIPaddress>/ippbx into the browser of any PC with Internet connectivity to access the admin login page for the PBX system.

Please note: If the end user already uses port 80, you will have to employ the 1 to 1 NAT port forwarding method to map a different external port (such as 8080) to the internal port 80. This is due to the PBX system's inability to modify the web access port.

==== Router Forwarding Interface ====
Example: DDWRT DDWRT is an open-source firmware compatible with a broad array of routers. It offers a user-friendly and relatively standardized configuration interface for setting up port forwarding. (An accompanying screenshot would showcase a router interface loaded with DDWRT Open Source firmware illustrating the configuration screen for Port Forwarding.)

Understanding and navigating these interfaces is essential for establishing and maintaining solid network configurations. Proper setup facilitates seamless communication between your network devices and the wider Internet, thereby enhancing your network's overall performance and security.

 

[[File:Router-PortForward.gif|none|Router-PortForward.gif]]

[[File:Router-PortRangeForwarding.gif|none|Router-PortRangeForwarding.gif]]
[[Category:Training]]

IPitomy SIP Trunk Configuration

2023-06-20T18:37:33Z

Tyler: /* Billing Portal */

__TOC__

E911

By default, when you get an IPitomy SIP Trunk it supports one CID and one Address. To ensure that E911 is getting the correct CID, make sure the SIP Provider has the correct Outbound CID set, and ensure that under your Emergency and Emergency Test routes (Call Routing=>Outgoing) have Disable EXT CID set to Yes. If you need to have more than one CID and Address due to multiple locations, you'll need to specify this when you are ordering the trunks.

[[File:EmergencyCIDOverrideDisabled.jpg|File:EmergencyCIDOverrideDisabled.jpg]]

'''TESTING WITH 933'''

It is of ''vital'' importance that you successfully test and verify the E911 service. Successful testing is required to begin the porting process.

Here are screen shots of both the Emergency 911 route and the Emergency Test 933 route. Please note that if there are phones at multiple physical locations, that you would need to create additional Emergency routes for each location that requires E911 service.  A Class of Service can be created and assigned to specific extensions to segregate each location.

[[File:E911 routes.jpg]]

== Migration ==

We have upgraded our infrastructure and have two new redundant regional servers we are using. If you have not done so already, the following link will take you to instructions on how to get this process started.

[[File:IPitomy SIP Trunk Migration.doc|File:IPitomy SIP Trunk Migration.doc]]

== Important Notes==

*All calls must go out these trunks as 11 digits, so be sure to add Prefix Digits to the trunks for the 7-Digits and 10-Digit routes.

*When naming the SIP trunk, do not use any spaces, hyphens, underscores, or special characters.

*When configuring the SIP ACL to allow IPitomy trunks to communicate to the PBX be sure to add the following: 52.5.220.123/32 and 54.200.236.200/32 for our regional redundant servers, East and West respectively.

*Moving forward, we will be building accounts only on the new regional redundant servers. Whether using IP Bound or Registration method, you'll need to add TWO SIP Providers, one for east and one for west. If using Registration method, you'll use the same Username and Password, but will need to change the HOST field in the registration string to reflect the correct server (sip-e1.ipitomy.com for East, sip-w1.ipitomy.com for West).

*Qualify must be set to 0 to ensure the trunks work consistently. We understand this means you can't look at the monitoring page to see the status of the trunks, but this is the best method. If we were monitoring and an options request packet was lost, the switch would not try sending calls till the next options request was received, meaning a period of lost calls. By not monitoring, all calls are attempted every time.

*When building Providers in the PBX for our SIP Trunks, we have one for East and one for West. Your DIDs should only be entered under one, and we recommend for that to be East since its the primary. If a call happens to come inbound over West, they will route to the correct location that is programmed for that DID under East, because in the background there is only one big table of DIDs.

== IP Bound Method ==

If the site that needs trunks has a static IP address, you will most likely be using the IP Bound Method to authenticate. This is going to be the easiest to set up.

::NOTE:: Qualify must be set to 0 to ensure the trunks work consistently. We understand this means you can't look at the monitoring page to see the status of the trunks, but this is the best method. If we were monitoring and an options request packet was lost, the switch would not try sending calls till the next options request was received, meaning a period of lost calls. By not monitoring, all calls are attempted every time. 
:: 

Please be sure to configure one SIP Provider for IPitomySIPEast, and one for IPitomySIPWest, to ensure you are fully taking advantage of our redundant infrastructure.

See the [[IP_PBX_Manual_Providers#IPitomy_SIP_Wizard | IPitomy SIP Wizard]] for a quick and easy way to configure the basics of your IPitomy SIP Trunks.

[[File:IPBound1.jpg|File:IPBound1.jpg]]

[[File:IPBound2.jpg|File:IPBound2.jpg]]

== Registration Method ==

If the site that needs trunks has a dynamic IP address, you will most likely be using the Registration Method to authenticate. While this takes a little more programming, it will work just as well. You should have documentation for the trunk listing the Host, Username, Password, and Domain to be used to fill in the variables in the example screenshot below: Please be sure to configure one SIP Provider for IPitomySIPEast, and one for IPitomySIPWest, to ensure you are fully taking advantage of our redundant infrastructure.

::NOTE:: Qualify must be set to 0 to ensure the trunks work consistently. We understand this means you can't look at the monitoring page to see the status of the trunks, but this is the best method. If we were monitoring and an options request packet was lost, the switch would not try sending calls till the next options request was received, meaning a period of lost calls. By not monitoring, all calls are attempted every time.

See the [[IP_PBX_Manual_Providers#IPitomy_SIP_Wizard | IPitomy SIP Wizard]] for a quick and easy way to configure the basics of your IPitomy SIP Trunks.

[[File:IpitomySIPRegistration1.jpg|IpitomySIPRegistration1.jpg]]

[[File:IPitomySIPTrunkRegistration2.jpg|IPitomySIPTrunkRegistration2.jpg]]

== Avaya Configuration ==

We received the following information from one of our dealers. To our understanding this will get IPitomy SIP Trunks working with an Avaya.

 [[File:IPItomy SIP Trunk Installation on Avaya IP Office.pdf|File:IPItomy SIP Trunk Installation on Avaya IP Office.pdf]]

Some additional notes from the dealer on what he's had to do to get our trunks working on Avaya:

*System/Network Topology/Stun Server Address – Use 216.93.246.18 (Counterpath Stun Server) then RUN STUN on the menu provided. Save settings.
*System/LAN 2/VoIP - Session Border Control (to block nefarious hacking) Change both RTP and NAT from default 49152-53246 to 46750-50750. Corresponding change needs to be completed in the Router / Firewall UDP Ports from 49152-53246 to 46750-50750 to address to LAN2 address of the PBX address of the PBX which in most cases is used to connect outside the system to internet. (The NIC on back of IP Office says WAN for LAN2)

== All Worx ==

We received the following images from one of our dealers. To our understanding this will get IPitomy SIP Trunks working with an Avaya.

[[File:AllworxConfig.zip|File:AllworxConfig.zip]]

== Grandstream FXS ==

In the rare instance the site does not have a VoIP PBX you may want to use a Grandstream FXS gateway to register to our SIP trunks, then connect the analog ports to the FXO ports on your traditional system. Here is what needs to be done to get the Grandstream registered to the IPitomy SIP Trunks.

*Connect the WAN port of the gateway to your network.
*Connect an analog phone to FXS port 1 and dial ***02 to find out the IP address of the Grandstream. You can also check in the router ARP table to find the IP.
*From the connected analog phone, dial ***129 to enable WAN HTTP Access.
*Set a static WAN IP, subnetmask, gateway, and DNS on the Basic Settings Page. This needs to be done as you will need to set up the standard remote SIP port forwards (5060, 10000-20000) in the router to this IP address.

[[File:Grandstream for IPitomy SIP 0.jpg|File:Grandstream for IPitomy SIP 0.jpg]]

*Note: The gateway has a static LAN IP of 192.168.2.1, if your networks is in the same subnet, you'll also need to modify the LAN DHCP Base IP so its on a different subnet or the device will not function correctly.
*Under Profile 1 you will need to configure the following:
**Set Primary SIP Server to the Domain provided for your Authenticated SIP Trunks
**Set the Outbound Proxy to the Host provided for your SIP Trunks
**Set NAT Transversal to Keepalive
**Set Hunt Group type to Linear
**[[File:Grandstream for IPitomy SIP 1.jpg|File:Grandstream for IPitomy SIP 1.jpg]]
**Set Use NAT IP to your sites static public IP
**[[File:Grandstream for IPitomy SIP 2.jpg|File:Grandstream for IPitomy SIP 2.jpg]]
**Set Add Auth Header On Initial REGISTER to Yes
**[[File:Grandstream for IPitomy SIP 3.jpg|File:Grandstream for IPitomy SIP 3.jpg]]
*Under FXO Ports configure the following:
**For Port 1, set SIP User ID and Authenticate ID to the Username provided for your Authenticated SIP Trunks
**For Port 1, set Password to the Password provided for your Authenticated SIP Trunks
***Note: Once saved, the field will blank out, this is just the way the gateway works so that it is not showing how long the password is.
**For Port 1, set Hunting Group to Active. Also set Enable Port to Yes.
**For the other ports (based on how many active lines you want to use) set Hunting Group to 1. This will allow one registration to roll over to additional lines. Be sure to set Enable Port to Yes for each port you want to use.
**[[File:Grandstream for IPitomy SIP 4.jpg|File:Grandstream for IPitomy SIP 4.jpg]]
*Update and Apply on each page, then Reboot the device when finished and you should be registered on FXS1 on the Status Page.
*Connected an analog phone, test inbound and outbound calls. Once its all working, you can connect the gateway to the FXO ports on your traditional system.

== Digium Gateway G100/G200/G400/G800 (IPitomy SIP Trunks to PRI) ==

When initially setting up the gateway, its default address is going to be [https://192.168.69.1 https://192.168.69.1] and its default username and password are both "admin"

1) Navigate to Configuration=>IP Configuration and define a static IP address for the gateway that is outside the scope of any DHCP servers on the network and save. Also be sure to set up a valid Gateway and DNS for the Digium.

NOTE: on versions prior to 2.3, there is a timing slip issue.  Before any further configuration, navigate to Maintenance=>Software Updates and ensure you are on the latest software version.  If not, run the automatic update.

http://wiki.ipitomy.com/images/5/57/Digium_g100_9.PNG
<div> </div>
2) Update your computer's IP and log back into the gateway.  Navigate to Configuration=>SIP Endpoints and press the Create SIP Endpoint button.  You will see the following screen.  Fill this out according to the details of your SIP account through us.  You will need the advanced options enabled for proper authentication.  If you are using IPitomy SIP trunks, you will need to make two endpoints.  Name them IPitomyEast and IPitomyWest, and use the appropriate addresses for each.  Save each.

[[File:Digium g100 1.PNG|File:Digium g100 1.PNG]]

[[File:Digium g100 2.PNG|File:Digium g100 2.PNG]]

3) Set the gateway up to emulate a PRI provider.  Navigate to Configuration=>T1/E1 Settings.  On the General Settings tab, simply ensure Use Internal Timing is enabled, and save:

[[File:Digium g100 3.PNG|File:Digium g100 3.PNG]]

4) Set up your PRI port (or ports if you have a gateway with more than one port and are using them with your PBX) as follows:

[[File:Digium g100 4.PNG|File:Digium g100 4.PNG]]
NOTE: Under Advanced Signalling, ensure "Enable Caller ID" is set to YES, or caller ID will not be passed properly.

5) You will need to create routing groups.  Navigate to Configuration=>Call Routing Groups, and press the button to create a call routing group.  You will need two groups at minimum.  One will have your SIP trunks as members, the other will have your PRI ports.  On the create routing group page, you simply name the group and check the boxes next to each member endpoint you wish to include in that group.

[[File:Digium g100 5.PNG|File:Digium g100 5.PNG]]

6) Navigate to Configuration=>Call Routing Rules.  By default the Digium gateway will come preconfigured with two rules: inbound_calls and outbound_calls.  Modify each rule so that outbound calls have From = your pbx group, and To = your trunk group.  Inbound calls should be exactly the opposite:  From = trunk group, To = pbx group.  These are the groups you created in the previous step.

[[File:Digium g100 6.PNG|File:Digium g100 6.PNG]]

[[File:Digium g100 7.PNG|File:Digium g100 7.PNG]]

[[File:Digium g100 8.PNG|File:Digium g100 8.PNG]]

7) On your pbx configuration (we will use an IPitomy PBX for example), you will need to enter the settings as if you were connect it directly to a PRI.  Using the settings you created above on the T1/E1 ports, it should look similar to this.  Most importantly, we set the gateway to provide timing above, so you will need to set the pbx to LISTEN for timing, not provide it:

[[File:Pbx 1.PNG|File:Pbx 1.PNG]]

[[File:Pbx 2.PNG|File:Pbx 2.PNG]]

You will use a standard PRI cable to connect the gateway ports to your pbx's PRI ports.  If everything is set up correctly as above, each port should flash yellow while it negotiates, and then turn to a solid green link light.

== Cisco Call Manager ==

We received the following Image and programming script from a dealer who has our SIP Trunks working on the users Cisco Call manager. This should help you to configure the same device yourself.

The guide goes over the script that needs to be used in the device.  In the script you will see the following that need to be replaced:

Replace <pbxIP> with the PBX Internal IP Address

Replace <1XXXXXXX...> with the numbers that your DIDs will have. Our understanding is if you had a DID 19413062200, and all of your DIDs started with 1941306, you'd enter 1941306....

[[File:Ipitomy Cisco Call Manager Guide.pdf|File:Ipitomy Cisco Call Manager Guide.pdf]]

The dealer also provided the following images of changes in the interface that needed to be done:

[[File:Image001.png|File:Image001.png]]

[[File:Image002.png|File:Image002.png]]

[[File:Image003.png|File:Image003.png]]

== Toshiba CIX ==

We received the following image and information from a successful configuration of our SIP trunks on a Toshiba CIX. It was important to note that the tech had to enter the DIDs in at 11 digits to get inbound calls to route correctly.  This was an IP Bound trunk.

[[File:IPitomySIPToshiba.png|File:IPitomySIPToshiba.png]]

== Cambium Networks FSX ATA ==

We received the following information from a dealer who configured IPitomy SIP Trunks to a Cambium ATA. Per the tech, their is a limitation that the number set as the login to the sip account will be the number sent as CID on outbound calls. See the following images and replace the <varialbles> with the correct information for our Authentication SIP Trunks.
[[File:CambiumFXSSIPPage.jpg|File:CambiumFXSSIPPage.jpg]][[File:CambiumFXSfxs1.jpg|File:CambiumFXSfxs1.jpg]]

Training:Process Review

2023-06-20T18:19:04Z

Tyler: /* IPitomy Cloud PBX */

== IPitomy PBX (On Premise) ==

==== '''Process Review''' ====
Preparation is critical before heading on-site to install a PBX, whether it's for an IPitomy cloud or on-premise solution. Thorough preparation reduces stress and ensures a smoother, more efficient installation process. This involves installing and testing a basic setup of the major components to ensure everything functions as expected.

==== Pre-Installation Steps ====
Thorough preparation is a key component of a successful installation. Complete the Site Survey and IPitomy Setup Worksheet, including contact information for key parties such as the ISP, IT Department, and Trunk Providers. If possible, set up as much as you can in the PBX before arriving on-site. Although it's not advisable to register the phones at this stage (since the IP address of the PBX may change), pre-configuring extensions, groups, menus, schedules, etc. reduces the time spent on-site during installation.

==== Matching the LAN ====
The first step of the installation process is configuring the PBX IP addresses to communicate on the network. If the network subnet is not 192.168.1.x, adjust the PBX to match the subnet of the LAN. There are two ways to configure the PBX IP address. After setting it up, it is recommended to reboot the PBX.

==== Using a Keyboard and Monitor ====
Connect a keyboard and monitor to the PBX and press ALT-F7. This will bring you to a screen that allows you to set the Static IP, Subnet Mask, Gateway, and DNS. Once all the values are set, select 'S' to save.

==== PC and Simple Network Setup ====
By default, you can access the PBX via 192.168.1.249/ippbx. Connect your PC and the PBX to a simple network, with only a switch between the two devices. Set your PC statically to 192.168.1.50 and log into the default IP address of the PBX. Once logged in, navigate to System => Networking and configure the Static IP, Subnet Mask, Gateway, and DNS.

==== Connecting to the Network ====
After configuring the correct IP addresses on the PBX, connect it to the customer network. Connect the PBX to the switch that will host the majority of the phones and avoid connecting it to the customer's router to prevent potential traffic bottlenecks.

==== SIP Localnet and External IP ====
Ensure the Localnet is properly configured under PBX Setup => SIP. The Localnet should match the LAN to allow phones to communicate with the PBX. The Localnet follows the pattern xxx.yyy.zzz.0, where x, y, and z match the PBX IP address, and the last octet is always zero. The subnet mask for the Localnet is typically 255.255.255.0. If remote SIP (Provider or Phones) is involved, enter the site's public IP address in the External IP field.

==== Access Control List ====
After setting the Localnet, configure the Access Control List (ACL) under System => Access Control => Access Control List. Click the 'Load Recommended Defaults' button to configure the basic ACL services (SIP, Call Manager, Local Manager, and TFTP) for devices within the Localnet to communicate with the PBX. If using a SIP provider, add <SIPTrunkIP>/32 as a rule to the SIP service in the ACL. For remote phones with static IP addresses, add them as well. If remote phones have non-static IP addresses, delete the entire SIP ACL Service and enable 'Log Watch & Ban'.

==== Registering Extensions ====
Start by registering two extensions and make test calls to ensure everything is functioning correctly. Verify if each phone can call the other, if there is two-way audio, and if there are any issues with call quality.

==== Testing Remote SIP ====
If the site plans to use remote phones or SIP trunks, install a remote phone to test if the router is handling NAT correctly. It's recommended to identify any issues at the beginning of the installation to allow time for router adjustments if necessary.

==== Testing with Softphones and Hardware Phones ====
Use a SIP softphone on a cell phone to test WAN extension registration to the PBX. Also, have other employees register physical SIP phones to WAN extensions on the PBX and test.

==== Configuring Trunks ====
Configure and test trunks early in the installation process. This allows the provider time to resolve any possible issues while you work on the rest of the installation. Add only one DID at this time to ensure the provider is sending the correct number of digits.

==== Thorough Testing ====
Thoroughly test the installation by setting up local extensions, remote extensions, and trunks. Ensure LAN phones can make and receive calls, DTMF works correctly, remote phones can make and receive calls, and trunks function properly. Verify there is two-way audio for LAN phones, WAN phones, and trunks, and check if DIDs are routing correctly.

==== Complete Configuration ====
After the basic installation has been tested and is functioning correctly, register the remaining phones to the PBX, add and configure the remaining DIDs, and thoroughly test the complete functionality. Check if Ring Group calls function as desired and if Menus route callers to the intended destinations.

==== Training ====
Once the system is installed and functioning as expected, begin training the end users. Many features will work similarly to their old system, but there may be new things to learn. Ensuring that end users are familiar with their phones and the PBX will result in satisfied customers.

== IPitomy Cloud PBX ==
IPitomy Cloud PBX Process Review Proper preparation is essential before initiating the setup of IPitomy's Cloud PBX solution. This includes creating a blueprint of the entire network setup, including SIP trunk providers and user extensions, which ensures an efficient and smooth installation process.

=== Pre-Installation Steps ===
Effective preparation is crucial to the successful implementation of a Cloud PBX solution. Completing the IPitomy Setup Worksheet, which includes important contact details for the ISP, IT Department, and Trunk Providers, is an essential first step. Pre-configuring extensions, groups, menus, schedules, etc., as much as possible before the implementation starts can significantly reduce the time required during the actual setup.

=== Cloud Configuration ===
Once the pre-installation steps are complete, configure the Cloud PBX to match the necessary specifications and requirements. This step includes setting up the appropriate firewall settings to ensure secure SIP communication, configuring necessary port forwarding rules if required, and preparing the system for integration with the network.

=== Web Interface Access ===
Access the IPitomy Cloud PBX interface via the provided URL. Once logged in, navigate through the system settings to configure the network settings, SIP trunk parameters, user extensions, and other necessary details as per the initial network blueprint.

=== Extension Setup ===
Set up the required extensions for users on the Cloud PBX system. This includes assigning extension numbers, setting up voicemail, configuring call forwarding rules if necessary, and providing the required credentials to the users for setting up their phones or softphones.

=== Remote SIP Testing ===
If the setup includes remote phones or SIP trunks, test them to ensure the router is handling Network Address Translation (NAT) correctly. It's essential to detect any potential issues early in the installation to allow time for necessary adjustments. If you are deploying remote phones at multiple locations you should always add an additional E911 end point for each location.

=== Softphone and Hardware Phone Testing ===
Use a SIP softphone on a mobile device to test the registration of WAN extensions to the Cloud PBX. Also, encourage employees to register their physical SIP phones to WAN extensions on the PBX and conduct tests to ensure they work properly.

=== Comprehensive Testing ===
Thoroughly test the system by setting up local extensions, remote extensions, and trunks. Confirm that all phones can make and receive calls, that DTMF is functioning correctly, and that all trunks are operating as they should. Check that there is two-way audio on all phones and that DIDs are routing correctly.

=== Final Configuration ===
After all tests confirm that the basic setup is functioning correctly, finalize the Cloud PBX configuration. This includes adding and configuring any remaining DIDs, setting up any additional features like call queues or IVRs, and testing the complete system to ensure everything is working as expected.

=== User Training ===
Once the Cloud PBX system is fully set up and functional, begin training the end users. While many features may be similar to their previous system, there will likely be new features and functions to learn. Ensuring that end users are comfortable with their phones and the new PBX system will result in satisfied customers.

[[Category:Training]]

IPitomy Communicator

2023-06-19T12:58:33Z

Tyler:

Note: This Guide will assist the user in Installing and Using the IPitomy Communicator Softphone.

We have two different soft phone installs. The “demo” version does not include the user list which shows all of the presence of all of the users on the system. This is intended for trial versions or for users who are on a shared instance of IPitomy in the Cloud. For obvious reasons, on a shared instance, it would not be good to display the presence of all the other users!

The full commercial version is called IPitomy Communicator.

This requires an available IPitomy Extension License as well as an IPitomy Soft Phone License. This does not have auto configuration, but is super simple to set up. When installing it, select IP620 as the device type for the extension so it will use an IPitomy license instead of an open license.

This also requires the generation of a single API Key. If your PBX already has one generated for the IPitomy Communicator, you don't need to do anything else. If not, simply navigate to Applications=>API Key, name it Softphone [one word, capital S], and click Create.  Be sure to enable it by checking the box labeled "Key Enable", then save and apply changes.

Links for SoftPhone:

Demo version: [http://relay.ipitomy.com/softcall_demo/publish.htm http://relay.ipitomy.com/softcall_demo/publish.htm]

Full version – IPitomy Communicator: [https://www.dropbox.com/s/6uhj12mwh8k7diu/CommunicatorSetup.zip?dl=0 <nowiki>[Here]</nowiki>]

== '''Ipitomy Softphone''' ==

'''Installation'''

 '''Installation Video'''

[https://www.youtube.com/watch?v=p5gwURMp4eo https://www.youtube.com/watch?v=p5gwURMp4eo]

 '''Park Button Video'''

 [https://www.youtube.com/watch?v=dPCVXiR6kf8 https://www.youtube.com/watch?v=dPCVXiR6kf8]

To Install the Softphone, Follow the URL above and click the Installer link. Installation will begin at that time. You should have adminstrative rights to the PC you are installing to prior to installation. Accept the default location for install unless you have some other directory you wish to install to.

'''Setup'''

Once isntalled open the program by clicking the program Icon. This should now be located on your Desktop as well as in the Program directory. The Program will open bringing up the user interface screen.

[[File:SoftPhone3.jpg|File:SoftPhone3.jpg]][[File:SoftPhone4.jpg|File:SoftPhone4.jpg]]

To Add an account Press the Settings Icon which is the little Gear Icon next to the Ipitomy Logo Bar. Select Sip Accounts and Add an Account

[[File:SoftPhone5.jpg|File:SoftPhone5.jpg]]

Select the Add Button and Enter the Following:

Display Name = Name of user

User Name = Extension Number

User Password = Voicemail Password

SIP Password = SIP Password for Extension

SIP Server = IP of PBX (Local IP is OK for a Local Extension only, however for a remote phone this would need to the Public IP of the Network the PBX resides upon)

Integration Port = Leave blank and the software uses 80 to verify its license with the PBX.  If the softphone is remote and an external port other than 80 is forwarded to the PBX, enter that as the Integration Port (eg, 8080 external forwarded to 80 internal)

All other information can be left at Default.

[[File:Communicator-SIP Acc.jpg|File:Communicator-SIP Acc.jpg]]

Be Sure to Click Enabled when you are returned to the account screen on the account you just created so it is made the live account

[[File:SoftPhone8.jpg|File:SoftPhone8.jpg]]

Click close and you should be returned to the User interface and your account should register to the PBX.

[[File:SoftPhone9.jpg|File:SoftPhone9.jpg]][[File:SoftPhone10.jpg|File:SoftPhone10.jpg]]

Your Softphone is now ready to be used.

'''Import Extensions from the PBX to Contacts.'''

To Import all Extension in the PBX into the Contact list select User List - Manage Users - Ipitomy and then Refresh

[[File:SoftPhone11.jpg|File:SoftPhone11.jpg]]

Select the PBX from the Host list

[[File:SoftPhone14.jpg|File:SoftPhone14.jpg]]

Click OK. The Softphone will now pull down an Extension list from the PBX which you configured.

[[File:SoftPhone13.jpg|File:SoftPhone13.jpg]]

Training:Site Survey

2023-06-12T20:51:00Z

Tyler: /* Site Survey */

= IPitomy Technical Training – Basic =

=== Site Survey ===

The key to avoiding unexpected challenges when transitioning numbers and executing cutover is a thorough site evaluation. This process scrutinizes both the readiness of the telephony and network for a premise or cloud IP PBX.

Whenever an IP PBX installation is due, ensure to fill out the [https://wiki.ipitomy.com/images/7/74/IPitomy_Setup_Worksheet.xls IPitomy Setup Guide.] It assists in structuring the PBX deployment by helping collate the information required to configure and prime the site application. The guide also serves as a tangible backup for the database configuration.

= Network & Telephony Readiness: =

== Service Type: SIP or Analog ==

*IPitomy recommends the use of IPitomy SIP trunks if reliable bandwidth is available
**Advantages are:
***High technical capabilities
***Native Integration [https://www.ipitomy.com/index.php/services/ipitomy-siptrunks (Purchase Directly from IPitomy])
***Reliable
***Instant Scalability (Additional Trunks/Lines can be added by IPitomy SIP Support)
***Mobility
***No Hardware or Software Costs
***Low Service & Maintenance Costs
***Easy to configure
***Guaranteed level of service from the provider

*Analog lines and PRI/T1 trunks at times are a necessary evil and should be used for fire alarms, elevators, security systems, etc. Check local state, city and county laws to determine if they are required. Build-in costs for additional testing and maintenance when you must use analog trunks. As part of the site survey you must know, for EACH Trunk circuit:
*Disadvantages of analog lines are:
**Low technical capabilities
**EOL
**High Cost
**Rarely an actual cable pair from CO to site
**Impacted by elements such as magnetic, radio, and weather interference
**No option for DID (Direct Inward Dial) feature
**EOL Announcement: No longer being maintained, supported or

== Phones & Extensions ==

*How many phones, what models and total extensions?
*Physically review each location where a telephone will be installed (Note: If there is an existing RJ45/Cat5 data drop/connection a phone may be used as a bridge connection to PC)
*Remote Employees will require additional programming, parts and service costs
*Analyze employees and their technical abilities; factor in training costs for users with little to no technical experience
*Prepare to train users – on-line training and documentation is available at [https://www.youtube.com/c/IPitomyOfficial IPitomy Documentation Repository].

= Network Readiness =

The network is the ''Backplane'' of the IP PBX. In old, obsolete telephony solutions (TDM, Switched Matrix) the backplane was designed and controlled by the PBX manufacturer – not so in IP Telephony. Networks must be setup and configured properly to allow for VoIP. Voice traffic should be given priority over all other network traffic types (QOS/COS). The benefits of voice on the network are numerous but voice traffic is not forgiving as is data traffic. RTP (Real Time Protocol) traffic (voice) cannot be interrupted without impacting voice connection performance. In rare cases where existing network performance is not up to VoIP standards (packet loss, latency, heavy load, etc), it may be beneficial to build out a second, physically separate voice network to protect and prevent voice quality from these issues.

== Cabling & WiFi – Is the correct cable in place? ==

*Cat5e or Cat6 cable is required
*Each location where an extension is to be placed must have an individual cable run to the location from the Switch location.
*Test each cable and the associated hardware (wall jack, connectors, and Patch Cables. Essentially ALL parts of the cable that will be used to connect the telephone to the Switch.)
*Are there any unused cables that are connected to the network – disconnect all unused hardware as they represent unknown transients on the network.
*Test WiFi signal and speed to determine if a WiFi phone can be deployed (must be connected to power outlet)

== WAN – ISP ==

*Who is the Internet Service Provider at the site?
*Should this change?
*Cost advantages can be offered in association with Internet Service Providers and IPitomy SIP Trunks.
*ISP Bandwidth & Performance (One VoIP Call uses ~90KBPS of bandwidth) <ul style="list-style-type:circle;">
<li>[http://speedtest.net/ Free Bandwidth and Speed Test]
<li>Keep in mind that ISP bandwidth testers are momentary snap-shots of the performance of your WAN (Internet) connection.</li></li>
<li>Several tests should be performed during the hours of operation at the site where the IP PBX will be installed.</li>
<li>IPitomy offers a [https://fs30.formsite.com/IPitomy/form44/index.html VoIP Qualification Test] for both on premise and cloud systems.</li>
</ul>

== Router ==

*What kind of Router is in use? Check [http://wiki.ipitomy.com/wiki/Router_Compatibility Compatibility Guide]
*Identify IP Address type:
**Static or Dynamic
*Who is Network Admin?
**If third party manages network, router, firewall, switches etc you will need their assistance and access to open & forward ports and enabling & disabling settings.
<ul style="list-style-type:circle;">
<li>Requirements
<li>Port Forwarding
</li>
<li>NAT (Network Address Translation)
<li>Disabling SIP ALG (Application Layer Gateway)
<li>ALG’s often contribute to one-way audio issues.
</li>*ALG’s often inhibit Remote Phone operation<li>Can DD-WRT be loaded onto the router on site?<ul style="list-style-type:circle;">
<li>We have found that the DD-WRT Linux-based router software generally NAT’s well.</li>
<li>[http://www.dd-wrt.com/ http://www.dd-wrt.com], then go to “Router Database” and search for the router using brand and model number informat</li>
</ul></li>
<li>Is a Sonicwall® present? If so, significant programming will be required to assure that no voice traffic (BOTH LAN and WAN traffic) is impeded.
<li>If using a Sonicwall® router/firewall;<ul style="list-style-type:circle;">
<li>Add cost to cover troubleshooting erratic telephone performance.</li>
<li>Consider installing a second Router in Gateway mode with its own Public IP to avoid Sonicwall® voice traffic issues.</li>
<li>Sonicwall® firewalls and routers with Sonicwall® firewalls add packet-evaluation algorithms that can “flag” voice traffic as offensive-behavior traffic and therefore impede voice connections.</li>
<li>Visit [[Router Info Sonicwall|Router Info: Sonicwall]] for IPitomy recommended programming information
</li>
</ul><ul style="list-style-type:circle;">
== Switch ==

The Switch is critical to network performance and quality voice connections. Patch cables connected to the switch must be of high quality and tested (as part of cabling) for each port of the Switch. We recommend use of Managed Switches with QOS capabilities.

Each time an IP PBX is to be deployed you should answer the following questions about the Switch(es) in place on site:

<ul><li>Are they managed (programmable)?</li>
<li>Are there enough ports to accommodate the number of telephones that will be installed?</li>
<li>Are the Switch ports POE (Power Over Ethernet)?<ul style="list-style-type:circle;">
<li>POE is desired for the “cleanest” installation since no power supply is required at each telephone location when using POE</li>
<li>POE is also desirable because it provides one point of backup (UPS on Switch supports telephones connected – they stay functional during power outage).</li>
</ul>

=== Functions of a Switch ===
*Analyze arriving packets
*Direct packet traffic to ports of the switch known to be connected to the packet-identified destination
*Apply traffic-rules to all traffic as specified in the managed table programming (QOS/COS)
*Can provide POE to power telephones from a central point

= Preliminary Verification =

VOIP is a packet-based technology as such it uses the network to deliver advanced communications services. The network will be shared with all network devices including the PBX and telephones. Assuring the voice packets have priority over data packets is essential to success. In some instances issues may arise that can cause network performance issues that can affect the proper delivery of communications services. Verifying that no problems exist on the network that would have a negative effect on voice communications is essential.

Options:
*Fully integrate into existing network IP Phones [IP330 shown] and IP PBX installed onto existing LAN with existing devices (printer, Laptop, PC, PC)

[[File:SiteSurvey1.gif|none]]
*Install separate Cat 5 cable plan for phones IP Phones (IP330 shown) and IP PBX installed using a separate network infrastructure; separate cable and separate data switch, the link between these two topologies is at the router.

[[File:SiteSurvey2.gif|none]]
*Separate network (voice / data) (separate router) IP Phones and IP PBX installed on a completely isolated network. Local traffic (LAN) between the networks are coupled using Router Bridge functions

[[File:SiteSurvey3.gif|none]]

Aids/tools:
*Network Analysis
**MicroConvergent (Third-party network analysis company) MC can help you pinpoint issues that elude you.
**SIP Trunk – many SIP providers provide SIP trunks for verification purposes.
**Perform a network analysis using available third-party tools or internally developed process.
*Obtain a SIP trunk from a recognized provider . Install this test SIP trunk onto your IPitomy demo system. Test the site using your demo IP PBX and IP Telephone and SIP trunk. This will give you a small window of the anticipated performance you may expect on site.

= Checklist =

Check off each of the above items for each site where an IPitomy IP PBX is to be installed. Seek to identify issues before cutover rather than after. The checklist will also help you to identify areas of concern that are not known network qualities… this is important if something goes wrong. A customer sign-off may be appropriate when a customer chooses not to follow recommendations.

Notes on Checklists:

<ul><li>Never assume that a network is performing at its highest efficiency</li>
<li>Always allow for corrective action to be taken if network performance does not sustain adequate voice traffic.<ul style="list-style-type:circle;">
<li>Be ready to bill your customer for corrective actions required that could not be known prior to the IP PBX application being installed.</li>
<li>SIP trunks and VoIP are internet-based and are therefore a non-guaranteed voice connection service due to fluctuations in bandwidth and WAN network performance.</li>
</ul>

= Common Sense Disclosure =

It is important to understand that the network performance is the responsibility of the customer. Any undisclosed issues that arise during installation that are network related must be resolved and may result in additional costs. This should be discussed in advance clearly stating who bears the responsibility for expenses related to optimizing network performance for VOIP.

Any changes made to the network should be disclosed prior to and after installation to avoid unnecessary service calls caused as a result of performance issues. For instance if a user plugs in a laptop with a fixed IP Address that is in conflict with any other network device (static or DHCP assigned) this device will cause network performance issues.

[[Category:Training]]

Training:Process Review

2023-06-12T20:27:29Z

Tyler:

=== IPitomy PBX (On Premise) ===

==== '''Process Review''' ====
Preparation is critical before heading on-site to install a PBX, whether it's for an IPitomy cloud or on-premise solution. Thorough preparation reduces stress and ensures a smoother, more efficient installation process. This involves installing and testing a basic setup of the major components to ensure everything functions as expected.

==== Pre-Installation Steps ====
Thorough preparation is a key component of a successful installation. Complete the Site Survey and IPitomy Setup Worksheet, including contact information for key parties such as the ISP, IT Department, and Trunk Providers. If possible, set up as much as you can in the PBX before arriving on-site. Although it's not advisable to register the phones at this stage (since the IP address of the PBX may change), pre-configuring extensions, groups, menus, schedules, etc. reduces the time spent on-site during installation.

==== Matching the LAN ====
The first step of the installation process is configuring the PBX IP addresses to communicate on the network. If the network subnet is not 192.168.1.x, adjust the PBX to match the subnet of the LAN. There are two ways to configure the PBX IP address. After setting it up, it is recommended to reboot the PBX.

==== Using a Keyboard and Monitor ====
Connect a keyboard and monitor to the PBX and press ALT-F7. This will bring you to a screen that allows you to set the Static IP, Subnet Mask, Gateway, and DNS. Once all the values are set, select 'S' to save.

==== PC and Simple Network Setup ====
By default, you can access the PBX via 192.168.1.249/ippbx. Connect your PC and the PBX to a simple network, with only a switch between the two devices. Set your PC statically to 192.168.1.50 and log into the default IP address of the PBX. Once logged in, navigate to System => Networking and configure the Static IP, Subnet Mask, Gateway, and DNS.

==== Connecting to the Network ====
After configuring the correct IP addresses on the PBX, connect it to the customer network. Connect the PBX to the switch that will host the majority of the phones and avoid connecting it to the customer's router to prevent potential traffic bottlenecks.

==== SIP Localnet and External IP ====
Ensure the Localnet is properly configured under PBX Setup => SIP. The Localnet should match the LAN to allow phones to communicate with the PBX. The Localnet follows the pattern xxx.yyy.zzz.0, where x, y, and z match the PBX IP address, and the last octet is always zero. The subnet mask for the Localnet is typically 255.255.255.0. If remote SIP (Provider or Phones) is involved, enter the site's public IP address in the External IP field.

==== Access Control List ====
After setting the Localnet, configure the Access Control List (ACL) under System => Access Control => Access Control List. Click the 'Load Recommended Defaults' button to configure the basic ACL services (SIP, Call Manager, Local Manager, and TFTP) for devices within the Localnet to communicate with the PBX. If using a SIP provider, add <SIPTrunkIP>/32 as a rule to the SIP service in the ACL. For remote phones with static IP addresses, add them as well. If remote phones have non-static IP addresses, delete the entire SIP ACL Service and enable 'Log Watch & Ban'.

==== Registering Extensions ====
Start by registering two extensions and make test calls to ensure everything is functioning correctly. Verify if each phone can call the other, if there is two-way audio, and if there are any issues with call quality.

==== Testing Remote SIP ====
If the site plans to use remote phones or SIP trunks, install a remote phone to test if the router is handling NAT correctly. It's recommended to identify any issues at the beginning of the installation to allow time for router adjustments if necessary.

==== Testing with Softphones and Hardware Phones ====
Use a SIP softphone on a cell phone to test WAN extension registration to the PBX. Also, have other employees register physical SIP phones to WAN extensions on the PBX and test.

==== Configuring Trunks ====
Configure and test trunks early in the installation process. This allows the provider time to resolve any possible issues while you work on the rest of the installation. Add only one DID at this time to ensure the provider is sending the correct number of digits.

==== Thorough Testing ====
Thoroughly test the installation by setting up local extensions, remote extensions, and trunks. Ensure LAN phones can make and receive calls, DTMF works correctly, remote phones can make and receive calls, and trunks function properly. Verify there is two-way audio for LAN phones, WAN phones, and trunks, and check if DIDs are routing correctly.

==== Complete Configuration ====
After the basic installation has been tested and is functioning correctly, register the remaining phones to the PBX, add and configure the remaining DIDs, and thoroughly test the complete functionality. Check if Ring Group calls function as desired and if Menus route callers to the intended destinations.

==== Training ====
Once the system is installed and functioning as expected, begin training the end users. Many features will work similarly to their old system, but there may be new things to learn. Ensuring that end users are familiar with their phones and the PBX will result in satisfied customers.

=== IPitomy Cloud PBX ===
[[Category:Training]]

Training:How does VOIP Work

2023-06-12T19:12:03Z

Tyler: /* Network Address Translation – NAT */

= How Does VoIP Work? =

This section provides an introductory overview of Voice over Internet Protocol (VoIP), a technology that enables the transmission of voice communications via IP networks.

VoIP works on the principle of audio sampling, where a computer records a sound (such as a human voice) at a high rate (typically at least 8,000 times per second) and converts these audio samples into digital data. Unlike traditional recording, where these samples are stored locally, VoIP sends these samples over an IP network to be played back on a different device.

The process of making VoIP function efficiently involves several key steps. Initially, the computer compresses the recorded sound samples to minimize the space they require, focusing particularly on voice frequencies. This compression and decompression process is handled by a tool known as a CODEC (compressor/decompressor). Numerous CODECs are available, and VoIP uses those optimized for voice compression, significantly reducing the bandwidth required compared to uncompressed audio.

After compression, the samples are grouped into larger units and inserted into data packets ready for transmission over the IP network, a process known as packetization. A typical IP packet can contain 10 or more milliseconds of audio, with 20 or 30 milliseconds being the most common.

A comparison could be made to sending postcards through traditional mail. Each postcard (packet) carries a limited amount of information. Sending a lengthy message would require multiple postcards (packets), and to ensure they can be assembled correctly at the destination, they are organized using a sequence number or similar mechanism.

[[File:How_voip_1.gif|alt=]]

Packets are sometimes delayed, just as with the postcards sent through the post office. This is particularly problematic for VoIP systems, as delays in delivering a voice packet means the information is too old to play. Such old packets are simply discarded, just as if the packet was never received. This is acceptable to a certain degree, as long as the assembled packets do not distort the sound. Too much delay will cause the sound to have less than desirable quality.

IP Devices generally measure the packet delay and expect the delay to remain relatively constant, though delay can increase and decrease during the course of a conversation. Variation in delay is called jitter.  Delay, itself, just means it takes longer for the recorded voice spoken by the first person to be heard by the user on the far end. In general, good networks have an end-to-end delay of less than 100ms, though delay up to 400ms is considered acceptable (especially when using satellite systems). Jitter can result in choppy voice or temporary glitches, so VoIP devices implement jitter buffer algorithms to compensate for jitter. Essentially, this means that a certain number of packets are queued before play-out and the queue length may be increased or decreased over time to reduce the number of discarded, late-arriving packets or to reduce "mouth to ear" delay. Such "adaptive jitter buffer" schemes are also used by a wide variety of devices that deal with variable delay.

 

== Jitter in Packet Voice Networks ==

Jitter is defined as a variation in the delay of received packets. At the sending side, packets are sent in a continuous stream with the packets spaced evenly apart. Due to network congestion, improper queuing, or configuration errors, this steady stream can become lumpy, or the delay between each packet can vary instead of remaining constant.

This diagram illustrates how a steady stream of packets is handled.
<div>[[File:HowVOIP2.gif|alt=|none|frame]]</div>When an IP device receives a Real-Time Protocol (RTP) audio stream for Voice over IP (VoIP), it must compensate for the jitter that is encountered. The mechanism that handles this function is the playout delay buffer. The playout delay buffer must buffer these packets and then play them out in a steady stream to the digital signal processors (DSPs) to be converted back to an analog audio stream. The playout delay buffer is also sometimes referred to as the de-jitter buffer.
This diagram illustrates how jitter is handled.

[[File:HowVOIP3.gif|alt=|none|frame]]

If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio. For losses as small as one packet, the DSP interpolates what it thinks the audio should be and no problem is audible. When jitter exceeds what the DSP can do to make up for the missing packets, audio problems are heard.

This diagram illustrates how excessive jitter is handled.
[[File:HowVOIP4.gif|alt=|none|frame]]

Video works in much the same way as voice. Video information received through a camera is broken into small pieces, compressed with a CODEC, placed into small packets, and transmitted over the IP network. This is one reason why VoIP is promising as a new technology: adding video or other media is relatively simple. Of course, there are certain issues that must be considered that are unique to video (e.g., frame refresh and much higher bandwidth requirements), but the basic principles of VoIP equally apply to video telephony.

Of course there is much more to VoIP than just sending the audio/video packets over the Internet. There must also be an agreed protocol for how computers find each other and how information is exchanged in order to allow packets to ultimately flow between the communicating devices. There must also be an agreed format (called payload format) for the contents of the media packets.

VoIP is implemented in a variety of hardware devices, including IP phones, analog terminal adapters (ATAs), and gateways. In short, a large number of devices can enable VoIP communication, some of which allow one to use traditional telephone devices to interface with the IP networks.

In a well performing network, VoIP calls should be as clear or clearer that and other type of audio transmissions.  VoIP calls are pure digitized sound.  Each audio packet contains the pure audio just exactly as it is spoken into the microphone. 

High definition voice contains a wider range of frequencies than typical voice transmissions and will deliver surprisingly good audio that contains a richer sound than most toll quality calls.

= VoIP Protocols =

The success of VoIP communication hinges on employing an appropriate set of protocols. Here, we'll discuss IPitomy SIP Trunking, the preferred protocol for all IPitomy devices currently in circulation as well as many other third-party industry offerings.

The Real-Time Protocol (RTP) is a standard used globally by nearly every device for transmitting audio and video packets between computers. RTP, guided by open standards outlined in various documents, manages issues like packet order and employs mechanisms such as the Real-Time Control Protocol to address delay and jitter.

Before media can flow between two devices, protocols are utilized to locate the remote device and negotiate the media flow methods. These crucial protocols are known as call-signaling protocols, with the Session Initiation Protocol (SIP) being the most widely used.

Advantages of IPitomy SIP Trunks

SIP, a text-based protocol, is relatively straightforward for machines to understand. Its easy readability facilitates troubleshooting by allowing the inspection of packet contents without needing to decompile the software entirely. SIP's operation mirrors that of email, making it familiar and intuitive. Its addressing methods are particularly similar. SIP calls offer undiluted digital voice quality, free of distortion, delay, or echo in its native environment. Instant Scalability: Given IPitomy's IP PBX platform, capacity can be easily adjusted via a simple process in the admin GUI. Interoperability with a wide range of devices provides users with choices far beyond what any single vendor can offer. International dialing is supported. User location is inconsequential; remote users can be deployed globally without losing direct connectivity. An adequate internet connection is the only requirement.

Understanding a SIP Call

A SIP call comprises a signaling component and a Voice component. The paths for signaling and actual voice transmission differ for each call. Setup and teardown signaling for all calls operate over port 5060. Voice transmissions are conducted within the range of ports 10,000 to 20,000. These are virtual ports integral to TCPIP communication.

Preconditions for a SIP Call

Before a SIP call can occur, SIP endpoints capable of finding and being found by other SIP endpoints are needed. We'll restrict our SIP examples to endpoints that register to an IPitomy IP PBX for this training guide. While peer-to-peer SIP calls are possible, most are facilitated via an IP PBX or soft switch for a PSTN-like ease of dialing. The advantage of an IP PBX is that it simplifies calling another endpoint, allows call information to be stored for reporting, and manages call routing to local and remote extensions. Users dial phone numbers and extension numbers as they would with a legacy PBX system. Although no longer recommended due to their End-Of-Life status, an IPitomy IP PBX still supports analog PSTN lines and T1/PRI cards. SIP endpoints must register with the PBX to be included in the PBX database. Once registered, they can dial phone numbers and receive calls from other endpoints. These endpoints can be phones on the Local Area Network (LAN) or any location on the internet.

Starting a SIP Call

To start a call, the endpoint sends an invite to the server requesting the other endpoint's availability. This occurs on the signaling port (port 5060). If the other endpoint is ready, it sends an acknowledgement back to the initiating phone, which then sends the call information instructing the other phone on the ports to commence the RTP (voice) session. The RTP session is opened using the communicated ports.

Call termination is initiated by one of the endpoints sending a "bye" message, causing the call to hang up. This is a simplified explanation of a SIP call's lifecycle. When the call occurs on the LAN, it bypasses the router. The PBX instructs the endpoints on the ports to connect the RTP (voice) stream.

For calls to remote phones, the PBX understands that the phone is beyond the firewall. Router port configuration becomes necessary for signaling and RTP traffic at this stage.

Signaling Port 5060

Proper port configuration enables port 5060 to be forwarded in the router to the PBX system's LAN IP address, allowing the PBX to send signals to remote phones and receive requests from them.

RTP Ports 10,000 – 20,000 Port Range Forwarding

Once call setup occurs via signaling on Port 5060, RTP is set up using a range of ports forwarded to the PBX LAN IP address. The Port Range Forwarding feature in the Router is used to forward the range of Ports 10,000 – 20,000 for this purpose.

Each call requires two ports for RTP - one for sending and one for receiving. This is organized by the initiating phone. The router ports are open from inside the firewall. The remote phone receives the information about which ports to use from the SIP packets.

'''Local Phone Diagram'''

[[File:5060.png|alt=]]

'''Remote Phone Diagram'''

[[File:Academy - RDP v5.gif|alt=]]

== Network Address Translation – NAT ==

TCP/IP, short for Transmission Control Protocol/Internet Protocol, is the underlying communication protocol used for data exchange on the internet. It leverages unique IP addresses to deliver data to the correct device on a network. The types and classes of IP addresses play a crucial role in how data is routed across the internet.

If you're online, odds are you're using Network Address Translation (NAT). This becomes increasingly likely given the growing internet user base. As of 2023, the internet is accessed by nearly 5 billion people worldwide, approximately 63% of the global population. This figure continues to rise, with nearly 200 million new users connecting in the year leading up to April 2023.

[[File:InternetUsers2019.png|alt=|700x700px]]

So what does the size of the Internet have to do with NAT? Everything! For a computer to communicate with other computers and Web servers on the Internet, it must have an IP address. An IP address (IP stands for Internet Protocol) is a unique 32-bit number that identifies the location of your computer on a network. Basically it works just like your street address: a way to find out exactly where you are and deliver information to you.

When IP addressing first came out, everyone thought that there were plenty of addresses to cover any need. Theoretically, you could have 4,294,967,296 unique addresses (232). The actual number of available addresses is smaller (somewhere between 3.2 and 3.3 billion) because of the way that the addresses are separated into Classes and the need to set aside some of the addresses for multicasting, testing or other specific uses.

With the explosion of the Internet and the increase in home networks and business networks, the number of available IP addresses is simply not enough. The obvious solution is to redesign the address format to allow for more possible addresses. This is being developed (IPv6) but will take several years to fully implement because it requires modification of the entire infrastructure of the Internet.

[[File:PNPN.png|alt=|700x700px]]

In our current internet landscape, dominated by IPv4 addressing, there's a finite number of unique IP addresses available. This limitation makes it challenging to provide every internet-connected device with its own IP address. The solution to this conundrum lies in the technique of Network Address Translation (NAT). The NAT process, employed by routers, enables a multitude of devices (like PCs, smartphones, and more) to share a single public IP address. This technique effectively extends the life of the current IPv4 addressing system until the broader implementation of IPv6, which promises a virtually limitless pool of IP addresses.

NAT operates by allowing the router to relay data to devices on the local area network (LAN) because it recognizes each device's unique internal IP address and Media Access Control (MAC) ID. When an external device needs to communicate with a device on the LAN via the internet, a specific route is required.

Consider the case of a remote IP phone initiating a call through a PBX system on the LAN. The phone needs to send packets to the PBX, and the router must be informed where to route these packets. This is achieved by forwarding port 5060 to the PBX on the LAN, meaning all traffic arriving at this port is directed to the PBX.

Once the call is established, the Real-Time Protocol (RTP) traffic is directed to designated ports for transmission and reception. These ports are assigned based on instructions in the SIP packets used in the call setup. Misconfiguration of port forwarding can cause issues, the most common of which is "one-way audio," typically resulting from improper configuration of the RTP ports in the router. Note that some routers support an Application Layer Gateway (ALG) functionality, which often hampers packet delivery, despite its seeming compatibility with SIP, and should thus be disabled.

Disruptions in the RTP stream can arise when voice packets cannot reach their intended destination due to router configuration errors or an inability of the router to correctly perform NAT operations. Some routers are outright incapable of NAT, making them incompatible with remote IP phones.

Having port forwarding enabled is critical, especially when considering remote access for maintenance, remote phones, and branch office connectivity. If a third party manages the router, all involved parties benefit from having these ports forwarded and the ALG disabled before the IP PBX is installed. Failure to follow these steps can lead to delays in implementation and should be factored in when providing price estimates to customers.

In IP telephony, TCP/IP and the SIP protocol are employed to generate pure digitized sound. Any distortions such as "static," echo, hiss, or hum are not introduced by the PBX but arise from analog elements or packet loss. To troubleshoot these issues, check the analog connections (like the handset and its cable) and conduct a packet loss test.

IP phones are intelligent and circuit-independent devices. You can simply unplug a problematic phone and plug it into a different Ethernet connection for troubleshooting. If the problem persists, try using a phone known to function properly on the problematic Ethernet connection. If the same issues arise, inspect the cables and connections. Ensure that Ethernet cables are not draped over fluorescent lights or close to other devices that could introduce distortion into the packet delivery process.

== Implementing Quality of Service (QOS) is Critical in Your VoIP Installation ==

Implementing Quality of Service (QoS) is crucial for ensuring optimal performance in your VoIP installation. Underestimating the importance of proper QoS configuration can result in a poor user experience and increased support costs. Let's explore the significance of QoS and how to set it up effectively.

'''What Does QoS Do?'''

QoS determines the priority of data packets on your Local Area Network (LAN). Since the available bandwidth on the LAN is shared by various applications, it is essential to prioritize voice packets for timely delivery. Voice packets are time-sensitive, and any interruption or delay can significantly degrade the audio quality of a call.

'''Voice Packets vs. Regular Data Packets'''

Voice packets are distinguished from regular data packets by a designated field in their header. This distinction allows the data switch to prioritize voice packets, ensuring they are not delayed or interrupted. In a network, data packets are typically delivered on a best-effort basis, utilizing available bandwidth. However, if the network becomes congested, voice packets may be momentarily blocked by other data packets. Even a brief delay can cause noticeable audio interruptions in phone calls. By prioritizing voice packets, you guarantee uninterrupted voice communication. Since voice traffic occupies a minimal percentage of the total bandwidth, prioritizing voice packets does not have a noticeable impact on other data packets.

For instance, consider a scenario where 10 people on the LAN are simultaneously downloading a 20-megabyte file. In a standard 100Base-T network, this heavy data traffic could potentially block all other data temporarily. By prioritizing voice packets over data packets, voice communication experiences no delay because the downloaded file makes room for voice packets with minimal or imperceptible delay to the ongoing downloads.

'''How to Set up QoS'''

QoS configuration is performed on the data switch. The IPitomy server uses specific settings to identify voice packets, which are set to CS3 by default. To ensure the highest priority for these packets, the data switch needs to be configured accordingly. Different switches employ various QoS labels, so you should determine the switch's specification to proceed. Since IPitomy utilizes the DSCP Class label, match that label in the switch to its highest priority setting (this could be a numerical value or "Highest," such as in the case of the Netgear FS728TP switch). It's important to ensure that no other devices on the network are using the same Class ID. If any other devices are utilizing it, either change their settings or modify the Class ID used by the IPitomy PBX under PBX Setup/SIP/Advanced. It is crucial to reserve the Class ID exclusively for voice traffic and not allocate it to other non-voice data devices.

Note: QoS can only be set on the LAN, specifically in the data switch(es). It is not relevant for the Wide Area Network (WAN) or internet traffic since those routes are determined by network hops that are beyond your control. However, in private WANs like MPLS, the network provider may have the capability to configure QoS for point-to-point connections.

VoIP (RTP) performs optimally when QoS is configured on the LAN. As a general rule, it is highly recommended to implement QoS for VoIP installations.

For further information on setting up QoS and its specific configuration for your system, please consult the relevant documentation or contact the IPitomy support team.

For more information on setting up QOS: [http://wiki.ipitomy.com/index.php/QOS_Setup_Guide Click Here].
[[Category:Training]]

Training:How does VOIP Work

2023-06-12T18:59:37Z

Tyler: /* VoIP Protocols */

= How Does VoIP Work? =

This section provides an introductory overview of Voice over Internet Protocol (VoIP), a technology that enables the transmission of voice communications via IP networks.

VoIP works on the principle of audio sampling, where a computer records a sound (such as a human voice) at a high rate (typically at least 8,000 times per second) and converts these audio samples into digital data. Unlike traditional recording, where these samples are stored locally, VoIP sends these samples over an IP network to be played back on a different device.

The process of making VoIP function efficiently involves several key steps. Initially, the computer compresses the recorded sound samples to minimize the space they require, focusing particularly on voice frequencies. This compression and decompression process is handled by a tool known as a CODEC (compressor/decompressor). Numerous CODECs are available, and VoIP uses those optimized for voice compression, significantly reducing the bandwidth required compared to uncompressed audio.

After compression, the samples are grouped into larger units and inserted into data packets ready for transmission over the IP network, a process known as packetization. A typical IP packet can contain 10 or more milliseconds of audio, with 20 or 30 milliseconds being the most common.

A comparison could be made to sending postcards through traditional mail. Each postcard (packet) carries a limited amount of information. Sending a lengthy message would require multiple postcards (packets), and to ensure they can be assembled correctly at the destination, they are organized using a sequence number or similar mechanism.

[[File:How_voip_1.gif|alt=]]

Packets are sometimes delayed, just as with the postcards sent through the post office. This is particularly problematic for VoIP systems, as delays in delivering a voice packet means the information is too old to play. Such old packets are simply discarded, just as if the packet was never received. This is acceptable to a certain degree, as long as the assembled packets do not distort the sound. Too much delay will cause the sound to have less than desirable quality.

IP Devices generally measure the packet delay and expect the delay to remain relatively constant, though delay can increase and decrease during the course of a conversation. Variation in delay is called jitter.  Delay, itself, just means it takes longer for the recorded voice spoken by the first person to be heard by the user on the far end. In general, good networks have an end-to-end delay of less than 100ms, though delay up to 400ms is considered acceptable (especially when using satellite systems). Jitter can result in choppy voice or temporary glitches, so VoIP devices implement jitter buffer algorithms to compensate for jitter. Essentially, this means that a certain number of packets are queued before play-out and the queue length may be increased or decreased over time to reduce the number of discarded, late-arriving packets or to reduce "mouth to ear" delay. Such "adaptive jitter buffer" schemes are also used by a wide variety of devices that deal with variable delay.

 

== Jitter in Packet Voice Networks ==

Jitter is defined as a variation in the delay of received packets. At the sending side, packets are sent in a continuous stream with the packets spaced evenly apart. Due to network congestion, improper queuing, or configuration errors, this steady stream can become lumpy, or the delay between each packet can vary instead of remaining constant.

This diagram illustrates how a steady stream of packets is handled.
<div>[[File:HowVOIP2.gif|alt=|none|frame]]</div>When an IP device receives a Real-Time Protocol (RTP) audio stream for Voice over IP (VoIP), it must compensate for the jitter that is encountered. The mechanism that handles this function is the playout delay buffer. The playout delay buffer must buffer these packets and then play them out in a steady stream to the digital signal processors (DSPs) to be converted back to an analog audio stream. The playout delay buffer is also sometimes referred to as the de-jitter buffer.
This diagram illustrates how jitter is handled.

[[File:HowVOIP3.gif|alt=|none|frame]]

If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio. For losses as small as one packet, the DSP interpolates what it thinks the audio should be and no problem is audible. When jitter exceeds what the DSP can do to make up for the missing packets, audio problems are heard.

This diagram illustrates how excessive jitter is handled.
[[File:HowVOIP4.gif|alt=|none|frame]]

Video works in much the same way as voice. Video information received through a camera is broken into small pieces, compressed with a CODEC, placed into small packets, and transmitted over the IP network. This is one reason why VoIP is promising as a new technology: adding video or other media is relatively simple. Of course, there are certain issues that must be considered that are unique to video (e.g., frame refresh and much higher bandwidth requirements), but the basic principles of VoIP equally apply to video telephony.

Of course there is much more to VoIP than just sending the audio/video packets over the Internet. There must also be an agreed protocol for how computers find each other and how information is exchanged in order to allow packets to ultimately flow between the communicating devices. There must also be an agreed format (called payload format) for the contents of the media packets.

VoIP is implemented in a variety of hardware devices, including IP phones, analog terminal adapters (ATAs), and gateways. In short, a large number of devices can enable VoIP communication, some of which allow one to use traditional telephone devices to interface with the IP networks.

In a well performing network, VoIP calls should be as clear or clearer that and other type of audio transmissions.  VoIP calls are pure digitized sound.  Each audio packet contains the pure audio just exactly as it is spoken into the microphone. 

High definition voice contains a wider range of frequencies than typical voice transmissions and will deliver surprisingly good audio that contains a richer sound than most toll quality calls.

= VoIP Protocols =

The success of VoIP communication hinges on employing an appropriate set of protocols. Here, we'll discuss IPitomy SIP Trunking, the preferred protocol for all IPitomy devices currently in circulation as well as many other third-party industry offerings.

The Real-Time Protocol (RTP) is a standard used globally by nearly every device for transmitting audio and video packets between computers. RTP, guided by open standards outlined in various documents, manages issues like packet order and employs mechanisms such as the Real-Time Control Protocol to address delay and jitter.

Before media can flow between two devices, protocols are utilized to locate the remote device and negotiate the media flow methods. These crucial protocols are known as call-signaling protocols, with the Session Initiation Protocol (SIP) being the most widely used.

Advantages of IPitomy SIP Trunks

SIP, a text-based protocol, is relatively straightforward for machines to understand. Its easy readability facilitates troubleshooting by allowing the inspection of packet contents without needing to decompile the software entirely. SIP's operation mirrors that of email, making it familiar and intuitive. Its addressing methods are particularly similar. SIP calls offer undiluted digital voice quality, free of distortion, delay, or echo in its native environment. Instant Scalability: Given IPitomy's IP PBX platform, capacity can be easily adjusted via a simple process in the admin GUI. Interoperability with a wide range of devices provides users with choices far beyond what any single vendor can offer. International dialing is supported. User location is inconsequential; remote users can be deployed globally without losing direct connectivity. An adequate internet connection is the only requirement.

Understanding a SIP Call

A SIP call comprises a signaling component and a Voice component. The paths for signaling and actual voice transmission differ for each call. Setup and teardown signaling for all calls operate over port 5060. Voice transmissions are conducted within the range of ports 10,000 to 20,000. These are virtual ports integral to TCPIP communication.

Preconditions for a SIP Call

Before a SIP call can occur, SIP endpoints capable of finding and being found by other SIP endpoints are needed. We'll restrict our SIP examples to endpoints that register to an IPitomy IP PBX for this training guide. While peer-to-peer SIP calls are possible, most are facilitated via an IP PBX or soft switch for a PSTN-like ease of dialing. The advantage of an IP PBX is that it simplifies calling another endpoint, allows call information to be stored for reporting, and manages call routing to local and remote extensions. Users dial phone numbers and extension numbers as they would with a legacy PBX system. Although no longer recommended due to their End-Of-Life status, an IPitomy IP PBX still supports analog PSTN lines and T1/PRI cards. SIP endpoints must register with the PBX to be included in the PBX database. Once registered, they can dial phone numbers and receive calls from other endpoints. These endpoints can be phones on the Local Area Network (LAN) or any location on the internet.

Starting a SIP Call

To start a call, the endpoint sends an invite to the server requesting the other endpoint's availability. This occurs on the signaling port (port 5060). If the other endpoint is ready, it sends an acknowledgement back to the initiating phone, which then sends the call information instructing the other phone on the ports to commence the RTP (voice) session. The RTP session is opened using the communicated ports.

Call termination is initiated by one of the endpoints sending a "bye" message, causing the call to hang up. This is a simplified explanation of a SIP call's lifecycle. When the call occurs on the LAN, it bypasses the router. The PBX instructs the endpoints on the ports to connect the RTP (voice) stream.

For calls to remote phones, the PBX understands that the phone is beyond the firewall. Router port configuration becomes necessary for signaling and RTP traffic at this stage.

Signaling Port 5060

Proper port configuration enables port 5060 to be forwarded in the router to the PBX system's LAN IP address, allowing the PBX to send signals to remote phones and receive requests from them.

RTP Ports 10,000 – 20,000 Port Range Forwarding

Once call setup occurs via signaling on Port 5060, RTP is set up using a range of ports forwarded to the PBX LAN IP address. The Port Range Forwarding feature in the Router is used to forward the range of Ports 10,000 – 20,000 for this purpose.

Each call requires two ports for RTP - one for sending and one for receiving. This is organized by the initiating phone. The router ports are open from inside the firewall. The remote phone receives the information about which ports to use from the SIP packets.

'''Local Phone Diagram'''

[[File:5060.png|alt=]]

'''Remote Phone Diagram'''

[[File:Academy - RDP v5.gif|alt=]]

== Network Address Translation – NAT ==

TCPIP is the protocol for sending data on the Internet.  It relies on unique IP addresses in order to get the proper data to the proper computer/device on the network.  There are several different types and classes of IP address. 

If you are reading this, you are most likely connected to the Internet and there's a very good chance that you are using Network Address Translation (NAT) right now!

The Internet has grown larger than anyone ever imagined it could be. Although the exact size is unknown, A total of '''5 billion''' people around the world use the internet today – equivalent to 63 percent of the world's total population. Internet users continue to grow too, with the latest data indicating that the world's connected population grew by almost 200 million in the 12 months to April 2022.

[[File:InternetUsers2019.png|alt=|700x700px]]

So what does the size of the Internet have to do with NAT? Everything! For a computer to communicate with other computers and Web servers on the Internet, it must have an IP address. An IP address (IP stands for Internet Protocol) is a unique 32-bit number that identifies the location of your computer on a network. Basically it works just like your street address: a way to find out exactly where you are and deliver information to you.

When IP addressing first came out, everyone thought that there were plenty of addresses to cover any need. Theoretically, you could have 4,294,967,296 unique addresses (232). The actual number of available addresses is smaller (somewhere between 3.2 and 3.3 billion) because of the way that the addresses are separated into Classes and the need to set aside some of the addresses for multicasting, testing or other specific uses.

With the explosion of the Internet and the increase in home networks and business networks, the number of available IP addresses is simply not enough. The obvious solution is to redesign the address format to allow for more possible addresses. This is being developed (IPv6) but will take several years to fully implement because it requires modification of the entire infrastructure of the Internet.

[[File:PNPN.png|alt=|700x700px]]

NAT Diagram – One Public IP Address is used by many Devices/Users Under the current IP addressing scenario (IPv4) there are a finite number of IP addresses available on the Internet.  There are not enough IP addresses available for each device to have their own unique IP address.  To solve this problem, all routers have the ability to send data to devices through a Network Address Translation (NAT) process.  This process allows a group of devices (like PC’s and Phones, etc.) to all share one Internet IP address.  This process has stretched out the usefulness of the current IP address scheme until the next numbering scheme (IPv6) is fully deployed.

NAT works by the router passing data to devices because it is aware of the address of the specific devices on the local area network.  The information you download to your PC comes directly to your PC because you have a unique internal IP address and a unique MAC ID. 

When a device from outside of the local area network, wants to communicate from the Internet to a device on the LAN, it needs a path to guide it to the specific device (like the PBX).  In the case of a remote IP phone, when the remote phone wants to make a call, it needs to send some packets to the PBX.  In order to do that, the router needs to be instructed on where to send the IP phone packets.  When port 5060 is forwarded to the PBX on the LAN, all traffic that comes in on port 5060 gets directed to the PBX.

Once the call is setup, the RTP traffic is directed to ports for sending and receiving.  These ports are determined through instructions in the call setup SIP packets.  If the port forwarding is not configured properly, the remote phone will not function properly.  The symptom most often associated to “one way audio” is almost always caused by improper configuration of the RTP ports in the router.  Some routers support Application Layer Gateway(ALG) functionality.  While this usually appears to be designed for SIP, it most often interferes with packet delivery and must be turned off.

It is easy to see how the RTP stream can be disrupted if the voice packets cannot reach the proper destination.  Sometimes this is caused by the router configuration.  Sometimes it can be the inability of the router to properly perform NAT functions.  Some routers are simply not capable of NAT and therefore will not work with remote IP phones.

It is essential to be in a position to have port forwarding enabled for remote access for maintenance, remote phones and branch office connectivity.  If a third party is in control of the router, it is in everyone’s best interest to have these ports forwarded and the ALG turned off and confirmed before the IP PBX is installed.  Failure to have these ports forwarded will result in implementation delays and must be a consideration when proposing a price for the end customer. 

IP Telephony over TCPIP using the SIP protocol produces pure digitized sound.  There are no functions inside the PBX to add sounds like “static”, echo, hiss or hum.  All of these sounds if present are produced in the analog world or are the result of packet loss.  In order to troubleshoot issues on a TCPIP packetized network, it is necessary to look for the solutions in the most likely places. 

If a customer complains of static, it is most often packet loss in an IP network or an analog entry point like a handset.  To identify the source of the problem, first check the analog connections e.g. handset, handset cable etc.  Try a known good handset and cord.  If that doesn’t solve the problem, run a test for packet loss. 

IP Phones are intelligent devices and are not dependent on a circuit.  It is easy to simply unplug the phone and plug it into another Ethernet connection.  If that fixes the problem, plug a known good phone into the Ethernet connection of the phone that had issues. If a known good phone is plugged in to the Ethernet connection and exhibits the same problem, check the cables and connections for problems.  Make sure the Ethernet cables are not draped over fluorescent lights are other devices that can induce distortion into the packet delivery process.

== Implementing Quality of Service (QOS) is Critical in Your VoIP Installation ==

Implementing QOS has huge benefits for your VoIP application. Don’t underestimate the importance of setting this up properly. Proper configuration can save customers from a difficult experience as well as keep your support costs down.

=== What Does QOS Do? ===

QOS sets the priority for data packets on your LAN. The LAN has packets from a diverse set of applications all traveling through a limited amount of bandwidth. Voice occupies a very small portion of the bandwidth. Since the voice packets are delivered in a time sensitive manner, it is important that they do not get interrupted or delayed. If they do, the audio quality on the call can deteriorate to a noticeable degree.

=== Voice Packets vs. Regular Data Packets ===

Voice Packets are distinguished from other data packets by a designation in the voice packet Header. This allows the data switch to know how to prioritize the individual packets to avoid delaying voice packets. Networks always try to deliver data on a best efforts basis. If there is bandwidth available, the data switch will try to pass all of the packets through as soon as it gets them using all of the available bandwidth. If this happens, the voice packets can be momentarily blocked by all of the other data. Even though this may only take a few seconds, it is enough of a delay to cause the phone call to experience audio interruptions as packets are delivered too late to be able to be used. By prioritizing the voice packets, you insure that the voice will never be interrupted. Since the voice is a very small percentage of total bandwidth, there is no noticeable effect on all of the other data packets.

An example would be that 10 people on the LAN are trying to download a 20 meg file at the same time. In a normal 100 base T network that could completely block all data traffic for a brief time. By prioritizing the voice packets to always take priority over the data packets, the voice is delivered without delay because the downloaded file makes room for the voice packets with little or no perceptible delay to the downloads.

=== How Do I Set up QOS? ===

QOS is set up in the data switch. The IPitomy server will have settings that it uses to identify the data packets. These settings are set to CS3 by default. The data switch will need to be configured to give the highest possible priority to these data packets. Switches use a variety of QOS labels so you will have to determine the scheme (specification) of the switch to be used. Since IPitomy uses the DSCP Class label, just match that label in the switch to the switch’s highest priority (this may be a digit or “Highest” as in the Netgear FS728TP). It’s important to know that no other devices on the network are utilizing that Class ID. If there are, change them or the IPitomy PBX under PBX Setup/SIP/Advanced. The Class ID used for voice traffic must not be used by other, non-voice data devices.

Note: QOS can only be set on the LAN [in the data switch(es)], it is not relevant on the WAN (Internet) since this media is routed by “hops” for which you have no control. The exception to this is private WAN’s like MPLS where the network provider may be able to configure QOS point-to-point.

VOIP (RTP) works best when QOS is set on the LAN. As a rule, always implement QOS.

For more information on setting up QOS: [http://wiki.ipitomy.com/index.php/QOS_Setup_Guide Click Here].
[[Category:Training]]

Training:How does VOIP Work

2023-06-12T18:49:34Z

Tyler:

= How Does VoIP Work? =

This section provides an introductory overview of Voice over Internet Protocol (VoIP), a technology that enables the transmission of voice communications via IP networks.

VoIP works on the principle of audio sampling, where a computer records a sound (such as a human voice) at a high rate (typically at least 8,000 times per second) and converts these audio samples into digital data. Unlike traditional recording, where these samples are stored locally, VoIP sends these samples over an IP network to be played back on a different device.

The process of making VoIP function efficiently involves several key steps. Initially, the computer compresses the recorded sound samples to minimize the space they require, focusing particularly on voice frequencies. This compression and decompression process is handled by a tool known as a CODEC (compressor/decompressor). Numerous CODECs are available, and VoIP uses those optimized for voice compression, significantly reducing the bandwidth required compared to uncompressed audio.

After compression, the samples are grouped into larger units and inserted into data packets ready for transmission over the IP network, a process known as packetization. A typical IP packet can contain 10 or more milliseconds of audio, with 20 or 30 milliseconds being the most common.

A comparison could be made to sending postcards through traditional mail. Each postcard (packet) carries a limited amount of information. Sending a lengthy message would require multiple postcards (packets), and to ensure they can be assembled correctly at the destination, they are organized using a sequence number or similar mechanism.

[[File:How_voip_1.gif|alt=]]

Packets are sometimes delayed, just as with the postcards sent through the post office. This is particularly problematic for VoIP systems, as delays in delivering a voice packet means the information is too old to play. Such old packets are simply discarded, just as if the packet was never received. This is acceptable to a certain degree, as long as the assembled packets do not distort the sound. Too much delay will cause the sound to have less than desirable quality.

IP Devices generally measure the packet delay and expect the delay to remain relatively constant, though delay can increase and decrease during the course of a conversation. Variation in delay is called jitter.  Delay, itself, just means it takes longer for the recorded voice spoken by the first person to be heard by the user on the far end. In general, good networks have an end-to-end delay of less than 100ms, though delay up to 400ms is considered acceptable (especially when using satellite systems). Jitter can result in choppy voice or temporary glitches, so VoIP devices implement jitter buffer algorithms to compensate for jitter. Essentially, this means that a certain number of packets are queued before play-out and the queue length may be increased or decreased over time to reduce the number of discarded, late-arriving packets or to reduce "mouth to ear" delay. Such "adaptive jitter buffer" schemes are also used by a wide variety of devices that deal with variable delay.

 

== Jitter in Packet Voice Networks ==

Jitter is defined as a variation in the delay of received packets. At the sending side, packets are sent in a continuous stream with the packets spaced evenly apart. Due to network congestion, improper queuing, or configuration errors, this steady stream can become lumpy, or the delay between each packet can vary instead of remaining constant.

This diagram illustrates how a steady stream of packets is handled.
<div>[[File:HowVOIP2.gif|alt=|none|frame]]</div>When an IP device receives a Real-Time Protocol (RTP) audio stream for Voice over IP (VoIP), it must compensate for the jitter that is encountered. The mechanism that handles this function is the playout delay buffer. The playout delay buffer must buffer these packets and then play them out in a steady stream to the digital signal processors (DSPs) to be converted back to an analog audio stream. The playout delay buffer is also sometimes referred to as the de-jitter buffer.
This diagram illustrates how jitter is handled.

[[File:HowVOIP3.gif|alt=|none|frame]]

If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio. For losses as small as one packet, the DSP interpolates what it thinks the audio should be and no problem is audible. When jitter exceeds what the DSP can do to make up for the missing packets, audio problems are heard.

This diagram illustrates how excessive jitter is handled.
[[File:HowVOIP4.gif|alt=|none|frame]]

Video works in much the same way as voice. Video information received through a camera is broken into small pieces, compressed with a CODEC, placed into small packets, and transmitted over the IP network. This is one reason why VoIP is promising as a new technology: adding video or other media is relatively simple. Of course, there are certain issues that must be considered that are unique to video (e.g., frame refresh and much higher bandwidth requirements), but the basic principles of VoIP equally apply to video telephony.

Of course there is much more to VoIP than just sending the audio/video packets over the Internet. There must also be an agreed protocol for how computers find each other and how information is exchanged in order to allow packets to ultimately flow between the communicating devices. There must also be an agreed format (called payload format) for the contents of the media packets.

VoIP is implemented in a variety of hardware devices, including IP phones, analog terminal adapters (ATAs), and gateways. In short, a large number of devices can enable VoIP communication, some of which allow one to use traditional telephone devices to interface with the IP networks.

In a well performing network, VoIP calls should be as clear or clearer that and other type of audio transmissions.  VoIP calls are pure digitized sound.  Each audio packet contains the pure audio just exactly as it is spoken into the microphone. 

High definition voice contains a wider range of frequencies than typical voice transmissions and will deliver surprisingly good audio that contains a richer sound than most toll quality calls.

= VoIP Protocols =

There are a number of protocols that may be employed in order to provide for VoIP communication services. In this section, we will focus on IPitomy SIP Trunking since it is the protocol of choice for all IPitomy devices now being deployed as well as third party manufacturers in the industry.

Virtually every device in the world uses a standard called Real-Time Protocol (RTP) for transmitting audio and video packets between communicating computers. RTP is defined by the open standards that are set using various standards documents. RTP also addresses issues like packet order and provides mechanisms (via the Real-Time Control Protocol, to help address delay and jitter.

Before audio or video media can flow between two devices, various protocols must be employed to find the remote device and to negotiate the means by which media will flow between the two devices. The protocols that are central to this process are referred to as call-signaling protocols, the most popular of which is Session Initiation Protocol (SIP).

Advantage of IPitomy SIP Trunks

*SIP is a simple protocol (at least for machines).  It is a text base protocol that is designed to be easily read and lends itself well to troubleshooting by being able to see what is in the packets without having to completely decompile the software,
*SIP works very similar to email; The addressing is very similar.
*SIP calls are pure digital voice.  In its native world, there is no distortion, no delay and no echo.
*Instant Scalability; Since IPitomy uses an IP PBX platform we can increase or decrease capacity with a simple, easy process in the admin GUI
*Many different devices can interoperate in a network providing a wide variety of choice for users that extends far beyond what any single vendor can provide. International dialing.
*Location of users is irrelevant; easily deploy remote users anywhere in the world without losing direct connectivity. The only requirement is an adequate internet connection.

Anatomy of a SIP Call

A SIP Call has a signaling component and a Voice component.  The signaling path is different from the actual voice transmission path.  The signaling and voice transmission are unique for each call.  All call setup and teardown signaling is over port 5060.  All voice transmissions are on a range of ports 10,000 to 20,000.  The ports are virtual ports and are simply part of the protocol for communicating over TCPIP. 

Setting the Stage

Before a SIP call can be placed, there needs to be SIP endpoints that have the ability to” find” and “be found by” other SIP endpoints in order to make and receive calls.  For the purposes of this training guide, we will limit our SIP examples to endpoints that register to an IPitomy IP PBX.  While it is possible to call from one SIP endpoint to another directly using a peer-to-peer method, most calls are facilitated through an IP PBX or soft switch to make dialing simple and easy just like a PSTN call.  The big advantage of having an IP PBX is that the user does not have to dial an IP address to call another endpoint and all of the call information can be stored for reporting etc. The IP PBX will handle routing all of the calls to their destination on the PSTN or to local and remote extensions.  Users simply dial phone numbers and extension numbers just like a legacy PBX system. Although not recommended due to their EOL an IPitomy IP PBX supports analog PSTN lines and T1/PRI cards. 

In order to be part of the PBX database, SIP endpoints register with the PBX.  Once they are registered, they have the ability to dial phone numbers and be called by other endpoints.  The registration can be from phones on the Local Area Network (LAN) or from anywhere on the Internet. 

Now that the phones are registered with the PBX, a call can be initiated or received.  To start the call, the endpoint sends an invite to the server to ask the other endpoint if it is available to take a call.  This takes place on the signaling port (port 5060).  If the other endpoint is ready to accept the call, it sends an acknowledgement back to the initiating phone.  The initiating phone then sends the call information telling the other phone which ports to commence the RTP (voice) session on.  The RTP session is opened using the ports communicated by the initiating call.

When the call is over, one of the endpoints sends a bye message and the call hangs up.  That is a pretty simple description of how SIP makes a call.  When the call is on the LAN it does not have to go through the router.  The PBX will handle telling the endpoints which ports to use to connect the RTP (voice) stream. 

When the call is to a remote phone, the PBX knows the phone is outside of the firewall.  This is when the router needs to have ports configured for signaling and RTP traffic. 

Signaling Port 5060

When the ports are properly configured, port 5060 is forwarded in the router to the PBX systems IP address on the LAN.  This allows the PBX to send signals to the remote phones as well as receive requests from them.

RTP Ports 10,000 – 20,000 Port Range Forwarding

Once the call is setup using the signaling on Port 5060, the RTP is setup using a range of ports that are forwarded to the PBX LAN IP address.  Using Port Range Forwarding in the Router, the range of Ports 10,000 – 20,000 is forwarded for this purpose.

Each call requires two ports for RTP; one for sending and one for receiving.  SIP sets this up from the initiating phone. The ports in the router are open from the inside of the firewall.  The phone on the far end receives the information on what ports to use in the SIP packets.

'''Local Phone Diagram'''

[[File:5060.png|alt=]]

'''Remote Phone Diagram'''

[[File:Academy - RDP v5.gif|alt=]]

== Network Address Translation – NAT ==

TCPIP is the protocol for sending data on the Internet.  It relies on unique IP addresses in order to get the proper data to the proper computer/device on the network.  There are several different types and classes of IP address. 

If you are reading this, you are most likely connected to the Internet and there's a very good chance that you are using Network Address Translation (NAT) right now!

The Internet has grown larger than anyone ever imagined it could be. Although the exact size is unknown, A total of '''5 billion''' people around the world use the internet today – equivalent to 63 percent of the world's total population. Internet users continue to grow too, with the latest data indicating that the world's connected population grew by almost 200 million in the 12 months to April 2022.

[[File:InternetUsers2019.png|alt=|700x700px]]

So what does the size of the Internet have to do with NAT? Everything! For a computer to communicate with other computers and Web servers on the Internet, it must have an IP address. An IP address (IP stands for Internet Protocol) is a unique 32-bit number that identifies the location of your computer on a network. Basically it works just like your street address: a way to find out exactly where you are and deliver information to you.

When IP addressing first came out, everyone thought that there were plenty of addresses to cover any need. Theoretically, you could have 4,294,967,296 unique addresses (232). The actual number of available addresses is smaller (somewhere between 3.2 and 3.3 billion) because of the way that the addresses are separated into Classes and the need to set aside some of the addresses for multicasting, testing or other specific uses.

With the explosion of the Internet and the increase in home networks and business networks, the number of available IP addresses is simply not enough. The obvious solution is to redesign the address format to allow for more possible addresses. This is being developed (IPv6) but will take several years to fully implement because it requires modification of the entire infrastructure of the Internet.

[[File:PNPN.png|alt=|700x700px]]

NAT Diagram – One Public IP Address is used by many Devices/Users Under the current IP addressing scenario (IPv4) there are a finite number of IP addresses available on the Internet.  There are not enough IP addresses available for each device to have their own unique IP address.  To solve this problem, all routers have the ability to send data to devices through a Network Address Translation (NAT) process.  This process allows a group of devices (like PC’s and Phones, etc.) to all share one Internet IP address.  This process has stretched out the usefulness of the current IP address scheme until the next numbering scheme (IPv6) is fully deployed.

NAT works by the router passing data to devices because it is aware of the address of the specific devices on the local area network.  The information you download to your PC comes directly to your PC because you have a unique internal IP address and a unique MAC ID. 

When a device from outside of the local area network, wants to communicate from the Internet to a device on the LAN, it needs a path to guide it to the specific device (like the PBX).  In the case of a remote IP phone, when the remote phone wants to make a call, it needs to send some packets to the PBX.  In order to do that, the router needs to be instructed on where to send the IP phone packets.  When port 5060 is forwarded to the PBX on the LAN, all traffic that comes in on port 5060 gets directed to the PBX.

Once the call is setup, the RTP traffic is directed to ports for sending and receiving.  These ports are determined through instructions in the call setup SIP packets.  If the port forwarding is not configured properly, the remote phone will not function properly.  The symptom most often associated to “one way audio” is almost always caused by improper configuration of the RTP ports in the router.  Some routers support Application Layer Gateway(ALG) functionality.  While this usually appears to be designed for SIP, it most often interferes with packet delivery and must be turned off.

It is easy to see how the RTP stream can be disrupted if the voice packets cannot reach the proper destination.  Sometimes this is caused by the router configuration.  Sometimes it can be the inability of the router to properly perform NAT functions.  Some routers are simply not capable of NAT and therefore will not work with remote IP phones.

It is essential to be in a position to have port forwarding enabled for remote access for maintenance, remote phones and branch office connectivity.  If a third party is in control of the router, it is in everyone’s best interest to have these ports forwarded and the ALG turned off and confirmed before the IP PBX is installed.  Failure to have these ports forwarded will result in implementation delays and must be a consideration when proposing a price for the end customer. 

IP Telephony over TCPIP using the SIP protocol produces pure digitized sound.  There are no functions inside the PBX to add sounds like “static”, echo, hiss or hum.  All of these sounds if present are produced in the analog world or are the result of packet loss.  In order to troubleshoot issues on a TCPIP packetized network, it is necessary to look for the solutions in the most likely places. 

If a customer complains of static, it is most often packet loss in an IP network or an analog entry point like a handset.  To identify the source of the problem, first check the analog connections e.g. handset, handset cable etc.  Try a known good handset and cord.  If that doesn’t solve the problem, run a test for packet loss. 

IP Phones are intelligent devices and are not dependent on a circuit.  It is easy to simply unplug the phone and plug it into another Ethernet connection.  If that fixes the problem, plug a known good phone into the Ethernet connection of the phone that had issues. If a known good phone is plugged in to the Ethernet connection and exhibits the same problem, check the cables and connections for problems.  Make sure the Ethernet cables are not draped over fluorescent lights are other devices that can induce distortion into the packet delivery process.

== Implementing Quality of Service (QOS) is Critical in Your VoIP Installation ==

Implementing QOS has huge benefits for your VoIP application. Don’t underestimate the importance of setting this up properly. Proper configuration can save customers from a difficult experience as well as keep your support costs down.

=== What Does QOS Do? ===

QOS sets the priority for data packets on your LAN. The LAN has packets from a diverse set of applications all traveling through a limited amount of bandwidth. Voice occupies a very small portion of the bandwidth. Since the voice packets are delivered in a time sensitive manner, it is important that they do not get interrupted or delayed. If they do, the audio quality on the call can deteriorate to a noticeable degree.

=== Voice Packets vs. Regular Data Packets ===

Voice Packets are distinguished from other data packets by a designation in the voice packet Header. This allows the data switch to know how to prioritize the individual packets to avoid delaying voice packets. Networks always try to deliver data on a best efforts basis. If there is bandwidth available, the data switch will try to pass all of the packets through as soon as it gets them using all of the available bandwidth. If this happens, the voice packets can be momentarily blocked by all of the other data. Even though this may only take a few seconds, it is enough of a delay to cause the phone call to experience audio interruptions as packets are delivered too late to be able to be used. By prioritizing the voice packets, you insure that the voice will never be interrupted. Since the voice is a very small percentage of total bandwidth, there is no noticeable effect on all of the other data packets.

An example would be that 10 people on the LAN are trying to download a 20 meg file at the same time. In a normal 100 base T network that could completely block all data traffic for a brief time. By prioritizing the voice packets to always take priority over the data packets, the voice is delivered without delay because the downloaded file makes room for the voice packets with little or no perceptible delay to the downloads.

=== How Do I Set up QOS? ===

QOS is set up in the data switch. The IPitomy server will have settings that it uses to identify the data packets. These settings are set to CS3 by default. The data switch will need to be configured to give the highest possible priority to these data packets. Switches use a variety of QOS labels so you will have to determine the scheme (specification) of the switch to be used. Since IPitomy uses the DSCP Class label, just match that label in the switch to the switch’s highest priority (this may be a digit or “Highest” as in the Netgear FS728TP). It’s important to know that no other devices on the network are utilizing that Class ID. If there are, change them or the IPitomy PBX under PBX Setup/SIP/Advanced. The Class ID used for voice traffic must not be used by other, non-voice data devices.

Note: QOS can only be set on the LAN [in the data switch(es)], it is not relevant on the WAN (Internet) since this media is routed by “hops” for which you have no control. The exception to this is private WAN’s like MPLS where the network provider may be able to configure QOS point-to-point.

VOIP (RTP) works best when QOS is set on the LAN. As a rule, always implement QOS.

For more information on setting up QOS: [http://wiki.ipitomy.com/index.php/QOS_Setup_Guide Click Here].
[[Category:Training]]

Training:Router

2023-05-17T22:28:30Z

Tyler:

<ul style="margin-left: 40px;">
<li>'''Introduction to Routers'''
<li>A router is a crucial component of any network. It serves as the digital 'postmaster,' managing the data traffic between your local network (LAN) and the vast expanse of the Internet. It performs this complex task through mechanisms like Network Address Translation (NAT), Port Forwarding, and by assigning IP addresses using Dynamic Host Configuration Protocol (DHCP). These features ensure a seamless, secure, and efficient routing of information from your devices to the Internet and back.

</li></li>
<li>'''Network Address Translation (NAT)'''
<li>In the digital world, IP addresses are a finite resource. Therefore, not every device connected to the Internet can have a unique public IP address. Instead, your local network has a distinct private subnet of IP addresses with a single public IP representing it online. Here's where NAT comes in handy.

When a device on your network wants to communicate with the Internet, the router uses NAT to map a unique port number to that device. The router then 'remembers' this mapping, ensuring that it knows where to direct any responses. Problems with NAT, such as inconsistent NAT, can often lead to connectivity issues. For instance, with remote phones, inconsistent NAT could lead to the device appearing unreachable or failing to receive calls.
</li></li>
<li>'''Dynamic Host Configuration Protocol (DHCP)'''
<li>Typically, routers also perform the role of a DHCP server. This means they're responsible for assigning IP addresses to devices on the network. In certain situations, a dedicated server on the network might handle DHCP instead. Understanding the configuration of DHCP during a site survey is crucial to prevent IP conflicts. You need to know how DHCP will be managed, the DHCP range, and a list of available static IP addresses for setting devices like PBX systems.

</li></li>
<li>'''Port Forwarding'''
<li>Port forwarding is how a router makes sure that incoming packets to specific ports are routed to the correct device on the LAN. For instance, remote phones initiate their communication by sending packets to port 5060. Hence, this port must be forwarded in the router to the PBX's internal static IP address.

</li></li>
<li>'''There are three main types of port forwarding:'''
* Single Port Forwarding: All incoming WAN traffic on a certain port is directed to a specific LAN IP via that port. For example, port 5060 externally can be forwarded to the PBX IP on port 5060.
* Port Range Forwarding: All incoming WAN traffic on a range of ports is directed to a specific LAN IP via that range of ports. For example, ports 10000 to 20000 externally can be forwarded to the PBX IP on ports 10000 to 20000.
* 1 to 1 NAT: All incoming WAN traffic on a certain port is directed to a specific LAN IP via a different port. This is usually employed when the required port is already in use. For example, if a user hosts their own webpage and port 80 is used, you can forward port 8080 externally to the PBX IP on port 80.

Proper router configuration is key to maintaining a stable and secure network. Misconfigurations can lead to unreachable devices, security vulnerabilities, or even complete network failure. So, understanding and managing these core router functions are essential skills for any network professional.
</li>
</ul>

 

Table 1: Single Port Forwarding

{| border="1" cellpadding="0" cellspacing="0" width="345"
|-
| Application Name
| Port
| Protocol
| To IP Address
|-
| Remote Administration
| 80
| TCP
| PBX Internal IP
|-
| SSH Support
| 22
| TCP
| PBX Internal IP
|-
| SIP
| 5060
| UDP
| PBX Internal IP
|-
| Branch Office
| 4569
| UDP
| PBX Internal IP
|}

 

Table 2: Port Range Forwarding

{| border="1" cellpadding="0" cellspacing="0" width="364"
|-
| Application Name
| Port
| Protocol
| To IP Address
|-
| RTP
| 10000-20000
| TCP & UDP
| PBX Internal IP
|}

 

Table 3: 1 to 1 NAT

{| border="1" cellpadding="0" cellspacing="0" height="94" width="523"
|-
| Application Name
| External Port
| Internal Port
| Protocol
| To IP Address
|-
| Alternate Remote Administration
| 8080
| 80
| TCP
| PBX Internal IP
|}

 

 

*'''Remote Access''' Remote access to devices like the PBX system enables network administrators or support staff to make configuration changes or troubleshoot issues from any location, not just on-site. This feature enhances the efficiency of network management and technical support services. To ensure maximum accessibility and visibility of the PBX system, it's recommended to forward port 80 (used for remote admin access) and port 22 (used for Secure Shell or SSH access) to the PBX's internal IP address. With this configuration, you can simply enter <code><publicIPaddress>/ippbx</code> in the browser of any internet-connected PC to reach the admin login for the PBX system. Note: If port 80 is already in use by the end user, you will need to use the 1 to 1 NAT port forwarding method to map a different external port (such as 8080) to the internal port 80. This is due to the inability to change the web access port on the PBX system.
*'''Example of Router Forwarding Interface: DDWRT''' DDWRT is an open-source firmware that's compatible with a wide range of routers. It provides a user-friendly and fairly standard configuration interface for setting up port forwarding. (Below would be a screenshot from a router interface loaded with DDWRT Open Source firmware showing the configuration screen for Port Forwarding.) Understanding and navigating these interfaces is crucial to setting up and maintaining robust network configurations. Proper setup ensures smooth communication between your network devices and the broader internet, enhancing your network's overall performance and security.

 

[[File:Router-PortForward.gif|none|Router-PortForward.gif]]

[[File:Router-PortRangeForwarding.gif|none|Router-PortRangeForwarding.gif]]
[[Category:Training]]

Training:Router

2023-05-17T22:27:56Z

Tyler:

<ul style="margin-left: 40px;">
<li>'''Introduction to Routers'''
<li>A router is a crucial component of any network. It serves as the digital 'postmaster,' managing the data traffic between your local network (LAN) and the vast expanse of the Internet. It performs this complex task through mechanisms like Network Address Translation (NAT), Port Forwarding, and by assigning IP addresses using Dynamic Host Configuration Protocol (DHCP). These features ensure a seamless, secure, and efficient routing of information from your devices to the Internet and back.

</li></li>
<li>'''Network Address Translation (NAT)'''
<li>In the digital world, IP addresses are a finite resource. Therefore, not every device connected to the Internet can have a unique public IP address. Instead, your local network has a distinct private subnet of IP addresses with a single public IP representing it online. Here's where NAT comes in handy.

When a device on your network wants to communicate with the Internet, the router uses NAT to map a unique port number to that device. The router then 'remembers' this mapping, ensuring that it knows where to direct any responses. Problems with NAT, such as inconsistent NAT, can often lead to connectivity issues. For instance, with remote phones, inconsistent NAT could lead to the device appearing unreachable or failing to receive calls.
</li></li>
<li>'''Dynamic Host Configuration Protocol (DHCP)'''
<li>Typically, routers also perform the role of a DHCP server. This means they're responsible for assigning IP addresses to devices on the network. In certain situations, a dedicated server on the network might handle DHCP instead. Understanding the configuration of DHCP during a site survey is crucial to prevent IP conflicts. You need to know how DHCP will be managed, the DHCP range, and a list of available static IP addresses for setting devices like PBX systems.

</li></li>
<li>'''Port Forwarding'''
<li>Port forwarding is how a router makes sure that incoming packets to specific ports are routed to the correct device on the LAN. For instance, remote phones initiate their communication by sending packets to port 5060. Hence, this port must be forwarded in the router to the PBX's internal static IP address.

</li></li>
<li>'''There are three main types of port forwarding:'''
* Single Port Forwarding: All incoming WAN traffic on a certain port is directed to a specific LAN IP via that port. For example, port 5060 externally can be forwarded to the PBX IP on port 5060.
* Port Range Forwarding: All incoming WAN traffic on a range of ports is directed to a specific LAN IP via that range of ports. For example, ports 10000 to 20000 externally can be forwarded to the PBX IP on ports 10000 to 20000.
* 1 to 1 NAT: All incoming WAN traffic on a certain port is directed to a specific LAN IP via a different port. This is usually employed when the required port is already in use. For example, if a user hosts their own webpage and port 80 is used, you can forward port 8080 externally to the PBX IP on port 80.

Proper router configuration is key to maintaining a stable and secure network. Misconfigurations can lead to unreachable devices, security vulnerabilities, or even complete network failure. So, understanding and managing these core router functions are essential skills for any network professional.
</li>
</ul>

 

Table 1: Single Port Forwarding

{| border="1" cellpadding="0" cellspacing="0" width="345"
|-
| Application Name
| Port
| Protocol
| To IP Address
|-
| Remote Administration
| 80
| TCP
| PBX Internal IP
|-
| SSH Support
| 22
| TCP
| PBX Internal IP
|-
| SIP
| 5060
| UDP
| PBX Internal IP
|-
| Branch Office
| 4569
| UDP
| PBX Internal IP
|}

 

Table 2: Port Range Forwarding

{| border="1" cellpadding="0" cellspacing="0" width="364"
|-
| Application Name
| Port
| Protocol
| To IP Address
|-
| RTP
| 10000-20000
| TCP & UDP
| PBX Internal IP
|}

 

Table 3: 1 to 1 NAT

{| border="1" cellpadding="0" cellspacing="0" height="94" width="523"
|-
| Application Name
| External Port
| Internal Port
| Protocol
| To IP Address
|-
| Alternate Remote Administration
| 8080
| 80
| TCP
| PBX Internal IP
|}

 

 

*'''Remote Access''' Remote access to devices like the PBX system enables network administrators or support staff to make configuration changes or troubleshoot issues from any location, not just on-site. This feature enhances the efficiency of network management and technical support services. To ensure maximum accessibility and visibility of the PBX system, it's recommended to forward port 80 (used for remote admin access) and port 22 (used for Secure Shell or SSH access) to the PBX's internal IP address. With this configuration, you can simply enter <code><publicIPaddress>/ippbx</code> in the browser of any internet-connected PC to reach the admin login for the PBX system. Note: If port 80 is already in use by the end user, you will need to use the 1 to 1 NAT port forwarding method to map a different external port (such as 8080) to the internal port 80. This is due to the inability to change the web access port on the PBX system. '''Example of Router Forwarding Interface: DDWRT''' DDWRT is an open-source firmware that's compatible with a wide range of routers. It provides a user-friendly and fairly standard configuration interface for setting up port forwarding. (Below would be a screenshot from a router interface loaded with DDWRT Open Source firmware showing the configuration screen for Port Forwarding.) Understanding and navigating these interfaces is crucial to setting up and maintaining robust network configurations. Proper setup ensures smooth communication between your network devices and the broader internet, enhancing your network's overall performance and security.

 

[[File:Router-PortForward.gif|none|Router-PortForward.gif]]

[[File:Router-PortRangeForwarding.gif|none|Router-PortRangeForwarding.gif]]
[[Category:Training]]

Training:Router

2023-05-17T22:25:54Z

Tyler:

<ul style="margin-left: 40px;">
<li>'''Introduction to Routers'''
A router is a crucial component of any network. It serves as the digital 'postmaster,' managing the data traffic between your local network (LAN) and the vast expanse of the Internet. It performs this complex task through mechanisms like Network Address Translation (NAT), Port Forwarding, and by assigning IP addresses using Dynamic Host Configuration Protocol (DHCP). These features ensure a seamless, secure, and efficient routing of information from your devices to the Internet and back.

'''Network Address Translation (NAT)'''

In the digital world, IP addresses are a finite resource. Therefore, not every device connected to the Internet can have a unique public IP address. Instead, your local network has a distinct private subnet of IP addresses with a single public IP representing it online. Here's where NAT comes in handy.

When a device on your network wants to communicate with the Internet, the router uses NAT to map a unique port number to that device. The router then 'remembers' this mapping, ensuring that it knows where to direct any responses. Problems with NAT, such as inconsistent NAT, can often lead to connectivity issues. For instance, with remote phones, inconsistent NAT could lead to the device appearing unreachable or failing to receive calls.

'''Dynamic Host Configuration Protocol (DHCP)'''

Typically, routers also perform the role of a DHCP server. This means they're responsible for assigning IP addresses to devices on the network. In certain situations, a dedicated server on the network might handle DHCP instead. Understanding the configuration of DHCP during a site survey is crucial to prevent IP conflicts. You need to know how DHCP will be managed, the DHCP range, and a list of available static IP addresses for setting devices like PBX systems.

'''Port Forwarding'''

Port forwarding is how a router makes sure that incoming packets to specific ports are routed to the correct device on the LAN. For instance, remote phones initiate their communication by sending packets to port 5060. Hence, this port must be forwarded in the router to the PBX's internal static IP address.

'''There are three main types of port forwarding:'''
* Single Port Forwarding: All incoming WAN traffic on a certain port is directed to a specific LAN IP via that port. For example, port 5060 externally can be forwarded to the PBX IP on port 5060.
* Port Range Forwarding: All incoming WAN traffic on a range of ports is directed to a specific LAN IP via that range of ports. For example, ports 10000 to 20000 externally can be forwarded to the PBX IP on ports 10000 to 20000.
* 1 to 1 NAT: All incoming WAN traffic on a certain port is directed to a specific LAN IP via a different port. This is usually employed when the required port is already in use. For example, if a user hosts their own webpage and port 80 is used, you can forward port 8080 externally to the PBX IP on port 80.

Proper router configuration is key to maintaining a stable and secure network. Misconfigurations can lead to unreachable devices, security vulnerabilities, or even complete network failure. So, understanding and managing these core router functions are essential skills for any network professional.
</li>
</ul>

 

Table 1: Single Port Forwarding

{| border="1" cellpadding="0" cellspacing="0" width="345"
|-
| Application Name
| Port
| Protocol
| To IP Address
|-
| Remote Administration
| 80
| TCP
| PBX Internal IP
|-
| SSH Support
| 22
| TCP
| PBX Internal IP
|-
| SIP
| 5060
| UDP
| PBX Internal IP
|-
| Branch Office
| 4569
| UDP
| PBX Internal IP
|}

 

Table 2: Port Range Forwarding

{| border="1" cellpadding="0" cellspacing="0" width="364"
|-
| Application Name
| Port
| Protocol
| To IP Address
|-
| RTP
| 10000-20000
| TCP & UDP
| PBX Internal IP
|}

 

Table 3: 1 to 1 NAT

{| border="1" cellpadding="0" cellspacing="0" height="94" width="523"
|-
| Application Name
| External Port
| Internal Port
| Protocol
| To IP Address
|-
| Alternate Remote Administration
| 8080
| 80
| TCP
| PBX Internal IP
|}

 

 

*'''Remote Access''' Remote access to devices like the PBX system enables network administrators or support staff to make configuration changes or troubleshoot issues from any location, not just on-site. This feature enhances the efficiency of network management and technical support services. To ensure maximum accessibility and visibility of the PBX system, it's recommended to forward port 80 (used for remote admin access) and port 22 (used for Secure Shell or SSH access) to the PBX's internal IP address. With this configuration, you can simply enter <code><publicIPaddress>/ippbx</code> in the browser of any internet-connected PC to reach the admin login for the PBX system. Note: If port 80 is already in use by the end user, you will need to use the 1 to 1 NAT port forwarding method to map a different external port (such as 8080) to the internal port 80. This is due to the inability to change the web access port on the PBX system. '''Example of Router Forwarding Interface: DDWRT''' DDWRT is an open-source firmware that's compatible with a wide range of routers. It provides a user-friendly and fairly standard configuration interface for setting up port forwarding. (Below would be a screenshot from a router interface loaded with DDWRT Open Source firmware showing the configuration screen for Port Forwarding.) Understanding and navigating these interfaces is crucial to setting up and maintaining robust network configurations. Proper setup ensures smooth communication between your network devices and the broader internet, enhancing your network's overall performance and security.

 

[[File:Router-PortForward.gif|none|Router-PortForward.gif]]

[[File:Router-PortRangeForwarding.gif|none|Router-PortRangeForwarding.gif]]
[[Category:Training]]

Training:Process Review

2023-05-17T22:08:02Z

Tyler:

'''Process Review'''

Preparation is critical before heading onsite to install a PBX, either for a cloud or on-premise IPitomy solution. It reduces stress and ensures a smoother, more efficient installation. This involves installing and testing a basic setup of the major components to make sure everything works as expected.

'''Pre-Install'''

Thorough preparation is a key component of a successful install. Your Site Survey and IPitomy Setup Worksheet should be completed, with the contact information for key parties like the ISP, IT Dept, Trunk Providers, etc. If possible, set up as much as you can in the PBX before heading onsite. While it's not advisable to register the phones just yet (since the IP address of the PBX may change), having your extensions, groups, menus, schedules, etc. pre-configured will reduce the time spent onsite during the installation.

'''Matching the LAN'''

The first step of your installation is configuring the PBX IP addresses to communicate on the network. If the network subnet isn't 192.168.1.x, you'll first need to adjust the PBX to communicate with the network. You can configure the PBX IP address to match the subnet of the LAN in two ways. After setting it up, it's recommended to reboot the PBX.

'''Keyboard and Monitor'''

With a keyboard and monitor connected, you can press ALT-F7. This will bring you to a screen that allows you to set your Static IP, Subnet Mask, Gateway, and DNS. Once all the values are set, select 'S' to save.

'''PC and Simple Network'''

By default, the PBX can be accessed via 192.168.1.249/ippbx. With your PC and PBX connected to a simple network (only a switch is needed between the two devices), and your PC set statically to 192.168.1.50, you can log into the default IP address of the PBX. Once logged in, navigate to System => Networking and change to Static IP, Subnet Mask, Gateway, and DNS.

'''Connecting to the Network'''

Once the PBX is configured with the correct IP addresses, connect it to the customer network. Be sure to connect the PBX to the switch that will host the majority of the phones. Avoid connecting the PBX to their router as this could create an unnecessary traffic bottleneck that could lead to issues down the line.

'''SIP Localnet and External IP'''

The Localnet defines what network is considered to be Local in terms of SIP communication. This setting can be found under PBX Setup => SIP. If the Localnet doesn't match the LAN, the phones won't be able to communicate with the PBX. The Localnet follows the pattern xxx.yyy.zzz.0, with x, y, and z matching the PBX IP address, and the last octet always being zero. The subnet mask for the Localnet will typically be 255.255.255.0. If the install involves any remote SIP (Provider or Phones), it's also advised to enter the public IP address for the site in the External IP field at this time.

'''Access Control List'''

Once the Localnet is set, it's recommended to set the ACL. This can be found under System => Access Control => Access Control List. Click the 'Load Recommended Defaults' button. This will configure the basic ACL services (SIP, Call Manager, Local Manager, and TFTP) to allow devices within the Localnet to communicate with the PBX. If you are using a SIP provider, add <SIPTrunkIP>/32 as a rule to the SIP service in the ACL. If there will be remote phones with static IP addresses, add those as well. For remote phones on non-static IP addresses, delete the entire SIP ACL Service and enable 'Log Watch & Ban'.

'''Register Two Extensions'''

Start by registering two extensions. After they're registered, make some test calls. This will help determine if everything is working correctly. Can each phone call the other? Is there two-way audio? Are there any issues with call quality?

'''Remote SIP'''

If the site plans to use any remote phones or SIP trunks, install a remote phone next to test if the router is handling NAT correctly. It's best to identify this at the start of the install instead of the end, as changes to the router might need to be made by a third party (like an IT Dept, Off-site IT, etc).

'''Softphone'''

One easy way to test would be to have a SIP softphone on your cell. Register this as a WAN extension to the PBX and test.

'''Hardware Phone'''

If you have other employees at your office, you can have them register a physical SIP phone to a WAN extension on the PBX and test.

'''Trunks'''

Configuring the trunks next and testing them will allow the Provider time to resolve possible issues while you work on the rest of the install. Waiting until the end to configure the trunks may prolong the cut-over time.

Add only one DID at this time to ensure the provider is sending the right number of digits.

'''Test'''

Thorough testing is vital. With a few local extensions, remote extensions, and trunks set up, you can get a good idea of how everything is working.

Can the LAN phones make and receive calls? Does DTMF work? Can Remote Phones make and receive calls? Can you make and receive calls via the Trunks? Is there two-way audio for LAN phones, WAN phones, and Trunks? Are DIDs routing correctly?

'''Configure the Rest'''

Once the basic install has been tested and is functional, register the remaining phones to the PBX, add and configure the remaining DIDs, and test the complete functionality. Do Ring Group calls function as desired? Do the Menus route the callers to the desired destinations?

'''Training'''

Once the system is installed and functioning as expected, the task of training the end user begins. Many features will work exactly like their old system did, but there will also be new things to learn. Ensuring the end user is up to speed on how to use their phones and PBX will result in a happier customer
[[Category:Training]]

Training:Install Preparation

2023-05-17T22:01:26Z

Tyler:

Before proceeding with the installation, it is crucial to identify the data switch that will be used, or recognize the existing switches at the site. We recommend utilizing a Level 2 managed switch, as it supports Quality of Service (QoS). Implementing QoS assures high-priority handling of voice traffic, thus improving call quality and overall performance. The switch you deploy is a fundamental component of your network's backbone.

When mapping out your network, consider the configuration and layout critically. Technicians often commit errors in network design, inadvertently creating unnecessary bottlenecks. For instance, if you have 20 phones connected to a 24-port switch, it would be optimal to connect the IPitomy PBX to this same switch. This setup ensures all voice network packets remain within the same hardware, optimizing network efficiency.

Another key consideration is the Power over Ethernet (PoE) option. PoE technology allows for the transmission of electrical power, along with data, over Ethernet cabling in a safe manner. The IEEE standard for PoE mandates Category 5 cable or above for high power levels, but Category 3 cable can suffice for lower power levels. Power is delivered in common mode over two or more of the differential pairs of wires found in Ethernet cables, sourced either from a power supply within a PoE-enabled networking device like an Ethernet switch or injected into a cable run with a midspan power supply.

The initial IEEE 802.3af-2003 PoE standard can provide up to 15.4 W of DC power (minimum 44 V DC and 350 mA) to each device, with 12.95 W guaranteed to be available at the powered device as some power loss occurs in the cable.

The subsequent IEEE 802.3at-2009 PoE standard, also referred to as PoE+ or PoE plus, can supply up to 25.5 W of power. This standard prevents a powered device from using all four pairs for power. However, some vendors have developed products claiming compatibility with the 802.3at standard, offering up to 51 W of power over a single cable by leveraging all four pairs in the Category 5 cable.

IPitomy telephones typically consume 7 watts of power or less, making them highly compatible with these PoE standards.

Before the standardization of PoE, various non-standard schemes were employed to deliver power over Ethernet cabling. Some of these schemes are still in use today, although adherence to IEEE standards is typically recommended for compatibility and safety reasons.

 [[File:NetworkTopology.gif|none]]

 

{| border="1" cellpadding="0" cellspacing="0"
|-
| style="height:56px" |
'''Speed [Mbit/s]'''

| style="height:56px" |
'''Distance [m]'''

| style="height:56px" |
'''Name'''

| style="height:56px" |
'''Standard / Year'''

| style="height:56px" |
'''Description'''

|-
| style="height:96px" |
1

| style="height:96px" |
100 (nominally)

| style="height:96px" |
[http://en.wikipedia.org/wiki/StarLAN StarLAN]

| style="height:96px" |
802.3e 1986[http://en.wikipedia.org/wiki/Ethernet_over_twisted_pair#cite_note-9 [9]]

| style="height:96px" |
Runs over four wires (two [http://en.wikipedia.org/wiki/Twisted_pair twisted pairs]) on telephone twisted pair or [http://en.wikipedia.org/wiki/Category_3_cable Category 3] cable. An active [http://en.wikipedia.org/wiki/Ethernet_hub hub] sits in the middle and has a port for each node. [http://en.wikipedia.org/wiki/Manchester_code Manchester coded] signaling.

|-
| style="height:96px" |
10

| style="height:96px" |
100 (nominally)

| style="height:96px" |
[http://en.wikipedia.org/wiki/LattisNet LattisNet]

| style="height:96px" |
(pre) 802.3i 1987

| style="height:96px" |
Runs over AT&T Premises Distribution System (PDS) wiring or four wires (two [http://en.wikipedia.org/wiki/Twisted_pair twisted pairs]) on telephone twisted pair or [http://en.wikipedia.org/wiki/Category_3_cable Category 3 cable].[http://en.wikipedia.org/wiki/Ethernet_over_twisted_pair#cite_note-syn-3 [3]][http://en.wikipedia.org/wiki/Ethernet_over_twisted_pair#cite_note-10 [10]]

|-
| style="height:144px" |
10

| style="height:144px" |
100 (nominally)[http://en.wikipedia.org/wiki/Ethernet_over_twisted_pair#cite_note-11 [11]]

| style="height:144px" |
10BASE-T

| style="height:144px" |
802.3i 1990

| style="height:144px" |
Runs over four wires (two [http://en.wikipedia.org/wiki/Twisted_pair twisted pairs]) on a [http://en.wikipedia.org/wiki/Category_3_cable Category 3] or [http://en.wikipedia.org/wiki/Category_5_cable Category 5 cable]. Star topology with an active [http://en.wikipedia.org/wiki/Ethernet_hub hub] or [http://en.wikipedia.org/wiki/Ethernet_switch switch] sits in the middle and has a port for each node. This is also the configuration used for 100BASE-T and gigabit Ethernet. [http://en.wikipedia.org/wiki/Manchester_code Manchester coded] signaling.

|-
| style="height:58px" |
100

| style="height:58px" |
100

| style="height:58px" |
[http://en.wikipedia.org/wiki/100BASE-TX 100BASE-TX]

| style="height:58px" |
802.3u 1995

| style="height:58px" |
[http://en.wikipedia.org/wiki/4B5B 4B5B] [http://en.wikipedia.org/wiki/MLT-3 MLT-3] coded signaling, [http://en.wikipedia.org/wiki/Category_5_cable CAT5] copper cabling with two twisted pairs.

|-
|
1000

|
100

|
[http://en.wikipedia.org/wiki/1000BASE%E2%80%91T 1000BASE‑T]

|
802.3ab 1999

|
[http://en.wikipedia.org/wiki/Pulse-amplitude_modulation PAM-5] coded signaling, At least [http://en.wikipedia.org/wiki/Category_5_cable Category 5 cable], with [http://en.wikipedia.org/wiki/Category_5e Category 5e] strongly recommended copper cabling with four twisted pairs. Each pair is used in both directions simultaneously.

|-
| style="height:58px" |
10 000

| style="height:58px" |
100

| style="height:58px" |
[http://en.wikipedia.org/wiki/10GBASE-T 10GBASE‑T]

| style="height:58px" |
802.3an 2006

| style="height:58px" |
Uses [http://en.wikipedia.org/wiki/Category_6a category 6a] cable.

|}

 

Installation:

When installing the system it is best to use the following outline:

*Create Extensions
**Import – extensions are created by importing a CSV file
**Auto Discovery  - PBX used to discover all phones on the network
**Auto Provision – PBX pairing function to install extensions
**Data Entry - one at a time
*Create Groups
**No limit to Groups
**Use for Department calling patterns
**Ring Strategies to be used
**Queue handling
*Create Menus
**Unlimited Menus
**Use for instructions/announcements
**Use for call routing
*Create Trunks
**SIP – Provider and SIP Configuration known and in-hand
**Hardware
***T-1, PRI
****DID destination
****Default Answering Destination
***Internal Gateway (PBX board for PSTN)
****Default Answering Destination
*Inbound Routing
** 
*Outbound Routing

Test as you go

*configure two phones and then test the connectivity, call between phones and voicemail to test
*configure the balance of the extensions
*When configuring a trunk, set it up and then test prior to full implementation.
**On SIP trunks, configure the trunk and then test inbound and outbound.
**On a PRI to check DID and make sure it is delivered correctly and only then program the balance of the DID’s.

 

When installing remote phones this method is also advantageous. To verify prior to rollout of all remote phones a single phone should be installed and tested to verify that the port configuration is all accurate.

 

Once all elements are configured and test correctly then you should proceed with the remainder of the install.

 
[[Category:Training]]

IPitomy Communicator

2023-04-24T15:34:47Z

Tyler:

Note: This Guide will assist the user in Installing and Using the IPitomy Communicator Softphone.

We have two different soft phone installs. The “demo” version does not include the user list which shows all of the presence of all of the users on the system. This is intended for trial versions or for users who are on a shared instance of IPitomy in the Cloud. For obvious reasons, on a shared instance, it would not be good to display the presence of all the other users!

The full commercial version is called IPitomy Communicator.

This requires an available IPitomy Extension License as well as an IPitomy Soft Phone License. This does not have auto configuration, but is super simple to set up. When installing it, select IP620 as the device type for the extension so it will use an IPitomy license instead of an open license.

This also requires the generation of a single API Key. If your PBX already has one generated for the IPitomy Communicator, you don't need to do anything else. If not, simply navigate to Applications=>API Key, name it Softphone [one word, capital S], and click Create.  Be sure to enable it by checking the box labeled "Key Enable", then save and apply changes.

Links for SoftPhone:

Demo version: [http://relay.ipitomy.com/softcall_demo/publish.htm http://relay.ipitomy.com/softcall_demo/publish.htm]

Full version – IPitomy Communicator: [https://www.dropbox.com/s/j67klcm8jk8bxev/CommunicatorSetup.zip?dl=0 <nowiki>[Here]</nowiki>]

== '''Ipitomy Softphone''' ==

'''Installation'''

 '''Installation Video'''

[https://www.youtube.com/watch?v=p5gwURMp4eo https://www.youtube.com/watch?v=p5gwURMp4eo]

 '''Park Button Video'''

 [https://www.youtube.com/watch?v=dPCVXiR6kf8 https://www.youtube.com/watch?v=dPCVXiR6kf8]

To Install the Softphone, Follow the URL above and click the Installer link. Installation will begin at that time. You should have adminstrative rights to the PC you are installing to prior to installation. Accept the default location for install unless you have some other directory you wish to install to.

'''Setup'''

Once isntalled open the program by clicking the program Icon. This should now be located on your Desktop as well as in the Program directory. The Program will open bringing up the user interface screen.

[[File:SoftPhone3.jpg|File:SoftPhone3.jpg]][[File:SoftPhone4.jpg|File:SoftPhone4.jpg]]

To Add an account Press the Settings Icon which is the little Gear Icon next to the Ipitomy Logo Bar. Select Sip Accounts and Add an Account

[[File:SoftPhone5.jpg|File:SoftPhone5.jpg]]

Select the Add Button and Enter the Following:

Display Name = Name of user

User Name = Extension Number

User Password = Voicemail Password

SIP Password = SIP Password for Extension

SIP Server = IP of PBX (Local IP is OK for a Local Extension only, however for a remote phone this would need to the Public IP of the Network the PBX resides upon)

Integration Port = Leave blank and the software uses 80 to verify its license with the PBX.  If the softphone is remote and an external port other than 80 is forwarded to the PBX, enter that as the Integration Port (eg, 8080 external forwarded to 80 internal)

All other information can be left at Default.

[[File:Communicator-SIP Acc.jpg|File:Communicator-SIP Acc.jpg]]

Be Sure to Click Enabled when you are returned to the account screen on the account you just created so it is made the live account

[[File:SoftPhone8.jpg|File:SoftPhone8.jpg]]

Click close and you should be returned to the User interface and your account should register to the PBX.

[[File:SoftPhone9.jpg|File:SoftPhone9.jpg]][[File:SoftPhone10.jpg|File:SoftPhone10.jpg]]

Your Softphone is now ready to be used.

'''Import Extensions from the PBX to Contacts.'''

To Import all Extension in the PBX into the Contact list select User List - Manage Users - Ipitomy and then Refresh

[[File:SoftPhone11.jpg|File:SoftPhone11.jpg]]

Select the PBX from the Host list

[[File:SoftPhone14.jpg|File:SoftPhone14.jpg]]

Click OK. The Softphone will now pull down an Extension list from the PBX which you configured.

[[File:SoftPhone13.jpg|File:SoftPhone13.jpg]]

Direct Inward Dialing

2023-04-11T14:33:37Z

Tyler:

== Direct Inward Dialing (DID) ==
A DID (Direct Inward Dialing) is a feature of VoIP phone systems that allows callers from the outside to dial directly into a specific phone extension within an organization, without having to go through a receptionist or operator. It is like having a personal phone number that is connected directly to your phone or device. This is different from traditional phone systems, where a receptionist or operator would have to manually transfer the call to the desired extension. DID numbers can be assigned to each individual within an organization, making it easier for clients and customers to reach the person they need to speak with directly. It also provides greater privacy, since personal phone numbers do not have to be shared with outside callers.

Having dedicated DIDs gives you more control and the ability to route incoming calls to any destination on your IPitomy system; Menus, Extensions, Schedules, Alerts, etc.

== Adding DIDs ==

Direct Inward Dialing

2023-04-11T14:30:28Z

Tyler:

Direct Inward Dialing

2023-04-11T14:24:07Z

Tyler: /* test */

Direct Inward Dialing

2023-04-10T18:37:23Z

Tyler: /* Direct Inward Dialing (DID) */

== Direct Inward Dialing (DID) ==

=== test ===

IP PBX Manual Intro

2023-04-10T18:36:37Z

Tyler: /* About the IPitomy IP PBX */

__NOTITLE__
{{IP_PBX_Manual|sortkey=Intro}}
[[Category:PBX Manual]]
<big>'''IPitomy PBX Manual Introduction'''</big>

= About the IPitomy IP PBX =

An IPitomy IP-PBX, or Internet Protocol Private Branch Exchange, is a phone system that uses internet protocol to route voice and multimedia communication within an organization or company. It replaces traditional PBX systems that use physical hardware to connect calls. With an IPitomy IP-PBX, calls can be routed over the internet and across multiple locations, making it ideal for companies with centralized and or distributed workforces. IPitomy offers both on Premise and Cloud (hosted) hosted IP-PBXs that are engineered to support; 1 to 500+ users. IPitomy systems give you instant scalability and flexibility for you to add or remove phone lines and features as needed, without having to invest in additional hardware. IPitomy also provides SIP trunking that is used to connect our IP-PBXs to the public switched telephone network, allowing for inbound and outbound calls to and from the system. IPitomy is also compatible with analog lines and T1 /PRI lines giving you the ability to implement our systems anywhere.

== Benefits of VoIP Technology ==

The IPitomy IP PBX can support any or all of these connectivity methods simultaneously or in any combination. Customers not quite ready to depend on VoIP providers for all of their business communications can start at their own pace and gain a comfort level, shifting to VoIP broadband providers at their own pace.

Benefits of VoIP technology include:

*'''One Wiring System:''' The system uses a single wiring system for telephones and dataall data and voice are on Local Area Network (LAN) Category 5 wiring.
*Software Based Upgrades: The IPitomy IP-PBX does not require any additional hardware to add extensions(phones) or advanced software features; ACD, Alerts, Scheduled Announcements, etc.
*'''Web-based Administration''': System administration is performed on the network through a Web-based administration program. The Web-Based Administration can be used locally or remotely from anywhere.
*'''Remote Users''': When calls are routed over the Internet, long distance charges can be avoided. In businesses with remote workers, these employees can stay logged into the office through a broadband connection at all times without incurring any additional charges. Remote users have all of the features of the local users. Remote users can be included in any ring groups, ACD (Automatic Call Distribution) Queues and other call routing schemes.
*'''Centralized System Features''': Every extension that is logged into the system is capable of receiving and originating calls. The use of system features such as voicemail, automated attendant and email are all centralized simplifying all support and maintenance.
*'''Reduced Costs''': VoIP system users can reduce cost in many areas of a business. VoIP telephony lowers the cost of support and maintenance costs, as well as, reducing telephony line costs by up to 50%.
*'''Simplifies Administration''': Moves, additions and changes are simple. The IPitomy IP PBX provides enhanced capabilities for users to make changes without incurring a service call.
*'''Investment Protection''': VoIP, and in particular, Session Initiation Protocol (SIP)-based VoIP products offer investment protection. The industry is rapidly moving toward Internet Protocol (IP) communications technologies. Older digital and analog technologies are becoming obsolete and are being replaced with IP-based products that will be around for a long time.

== IPitomy IP PBX Features ==

Understanding the IPitomy IP PBX’s architecture and how it works will make installing the system simple.

The IPitomy IP PBX is an all-in-one business communications system. This powerful system includes a complete suite of business communication applications in one appliance:

*Fully-featured Business Phone System
*Automated Attendant and Interactive Voice Response (IVR)
*Enhanced Call Distribution
*Enhanced Voice Messaging System with Unified Messaging
*Meet-me Conference Application
*Built-in Music on Hold
*Call Queuing for Inbound Calls
*Find Me/Follow Me
*Remote Extensions
*Browser-based Administration
*[[Branch Offices]]
*[[ACD|Automatic Call Distribution (ACD)]]
*[[Call Recording]]
*[[Routing Ops|Advanced Inbound Routing]]
*[[Alerts]]
*[[IPPBX IMM Scheduled Calls|Scheduled Calling]]
*[[Room Management System|Hospitality PMS (Property Management Integration) Integration & Room Management]]
*[[Q Manager|QManager]]
*[[IPitomy Cloud Connect|Mobile]] & [[IPitomy Communicator|Desktop]] Softphones

The IPitomy IP PBX’s administration menus are a series of Web pages accessible from a Web browser. To the left of the Menu is a navigation bar that allows users to click on and administer each section of the system. Administration of the IPitomy IP PBX is simple and intuitive. The system is designed with seven primary areas of functionality and security:

*'''System''': System setup consists of network configuration settings.
*'''Providers''': Providers are sources of PSTN and VoIP connectivity. Providers are the lines that handle all incoming and outgoing calls. All VoIP and traditional telephone providers are setup here. DID numbers are also entered here.
*'''Destinations''': Destinations are places where calls are routed in the system: extensions, groups of extensions, automated attendants, conferences, and voicemail.
*'''Call Routing''': These settings route inbound calls to specific destinations within the system, and send outbound calls over specific local, long distance, international, and emergency routes.
*'''PBX Setup''': These settings globally configure PBX timers, voice messaging, and other system features.
*'''Reporting''': These reports display system usage, monitor activity, and provide diagnostic information.
*'''Diagnostics''': Additional diagnostic and testing options.
*'''Security Features''': Log, watch, and ban. Access Control Lists.

{| class="wikitable"
|-
| <center>Feature</center>
| <center>Description</center>
|-
| [http://wiki.ipitomy.com/wiki/IPPBX_Manual_System_Admin_Adding_and_Deleting_Extensions Extensions] 
| Extensions are telephones. A telephone can be an IP (SIP) telephone or a Softphone. Calls are routed to an extension where people answer them. In the IPitomy IP PBX, an extension can be located in an office or outside the office where a broadband connection is used.
|-
| [http://wiki.ipitomy.com/wiki/IP_PBX_Manual_Destinations_Groups#Ring_Group_Examples Groups ]
| Groups are a set of extensions. Once a group is created, extensions can be designated as members of the group. This is accomplished by selecting group members from a drop- down list. Calls can be routed to groups via inbound routing.
|-
| [http://wiki.ipitomy.com/wiki/IP_PBX_Manual_Destinations_Menus#Menus Menus (Automated Attendant) ]
|
To create an automated attendant use the system’s Menus feature. The Menus feature allows you to route calls to a destination in the system like a group, extension or another menu.

Call Destinations are selected from a drop-down list for each corresponding key-pad digit a caller must select to get to their chosen destination. A Menu must have a Menu Prompt. This is a recording that identifies for callers the destinations they may choose. For example, a Menu Prompt might offer callers the option to press “1” for Sales, “2” for Accounts Receivable or other digits for another department.

|-
| [[Menu Management|Menu Management ]]
| This feature allows the user to administer, update, and change the Menu (Auto Attendant) remotely using just a telephone (with DTMF dial capability).
|-
| [[Unified Messaging|Voicemail and Unified Messaging ]]
| When an extension is created, a voicemail box for that extension is also created. A voicemail box allows a caller to leave a message if a person is not available at the extension. When dialing into a mailbox for the first time, a user should record their name and a mailbox greeting. The name is used in the company’s dial-by-name directory when selected from the auto attendant (Menus). The greeting is played when they are not available to take a call and a caller reaches their mailbox.
If an email address is included in the Extension page, you can configure Unified Messaging and a copy of the voicemail message will be emailed as a .Wav file to the users email account. This message can then be listened to on a PC.

|-
| [[Directory Listing|Directory ]]
| The system has a dial-by-name directory. This option may be part of the automated-attendant. When this option is selected, a caller dials the first three letters of the last / first name of the party they would like to reach. Names that match these three letters are played and the caller selects the extension to which they want to be transferred. Names are stated in the directory as they have been recorded by users in their voicemail box, and spelt out if they have not recorded their name.
|-
| [[Direct Inward Dialing (DID) Numbers|Direct Inward Dialing (DID) Numbers ]]
| A Direct Inward Dialed (DID) number is a telephone number assigned by a service provider (i.e., T1 line, PRI or VoIP). DIDs allow direct routing of a call to a destination within the system. You can route to any destination available on the PBX.
|-
| Conferencing (Meet Me) 
| A Meet-me Conference is an extension on the system used for conference calls. Participants can access a conference by dialing the designated Meet-me Conference extension. Routing callers to a Meet-me Conference can be accomplished by using a DID, a menu, or simply transferring callers to the conference extension.
|-
| Follow-Me 
| This feature allows the PBX to try and find users who are not at their desk. It can be configured to call their cell phones, house phones, or other extensions in the PBX. Once answered, the user can accept the call, or refuse it. Unhandled calls return to the PBX to leave a message at the original extension’s voicemail.
|-
| Forwarding Gateway 
| Mobility has become a part of everyday life for most people. System users need to be able to take calls anywhere. The IPitomy IP PBX has the ability to forward calls. Users can turn call forwarding “on” and “off” while in the office or away from the office by using a touch-tone key pad. This is set up in the edit Extensions page, but can be modified from any phone, including a cell phone. Modifying forward settings remotely requires the automated attendant (Menus) option to be programmed.
|-
| Cascading Message Notification 
| This feature works using the same methods as FollowMe, but pertains to voicemail messages. When configured, if an extension gets a new voicemail, you will be able to send the voicemail message to a variety of numbers (Destinations), define the order in which to send the message and can be set to make the system to notify you that a new message was received. Additionally, you can add or remove extensions to the list of recipients when a broadcast message is sent.
|-
| Voicemail Gateway 
| Using either a Menu or a DID, users can call in from any telephone and check messages. The voicemail gateway allows users to dial a pre-defined digit from a touch- tone key pad on any phone to retrieve their messages.
|-
| Branch Offices 
| Branch offices can be created to allow multiple PBXs to route calls to each other. Branch office extensions can be transferred to, placed in ring groups, or selected as menu destinations.
|-
| Automatic Call Recording 
| A licensed feature that allows you to set up for automatic recording of calls, inbound via Ring Groups or outbound via Outbound Routes.
|-
| Advanced Inbound Routing 
| A licensed feature that allows for calls to be routed inbound based on a number of options, including inbound CID or digits entered at a prompt.
|-
|Alerts
|A licensed feature: That can be set up to allow notifications to be sent out when a certain number is dialed as an outgoing call, such as 911.
|-
|Scheduled Calling
|A licensed feature that allows for periodic automated calls to be made by the PBX. This can be used for announcements, bells, alarms etc.
|-
|Hospitality & Room Management
|A licensed feature which allows you to specify PBX extensions as hotel guest room extensions, and manage the rooms associated with them
|-
|QManager
|A licensed feature: Q Manager is a '''Windows application''' that allows users to monitor and interact with calls
|-
|Softphones
|A licensed feature: IPitomy offers mobile and desktop softphones that give you the ability to instantly deploy remote/work from home or new extensions
|}

''Table 1IP PBX Features''

Direct Inward Dialing

2023-04-10T18:36:02Z

Tyler: Created page with "== Direct Inward Dialing (DID) =="

== Direct Inward Dialing (DID) ==

IP PBX Manual Intro

2022-09-06T13:39:59Z

Tyler: /* Benefits of VoIP Technology */

__NOTITLE__
{{IP_PBX_Manual|sortkey=Intro}}
[[Category:PBX Manual]]
<big>'''IPitomy PBX Manual Introduction'''</big>

= About the IPitomy IP PBX =

The IPitomy IP PBX is a powerful business communications platform. It is a pure IP PBX designed to use IP networks for voice calls. Engineered to support from 10 to 500 users, the system will work with analog lines and T1 /PRI lines for traditional Public Switched Telephone Network (PSTN) connectivity. In addition to traditional telephone lines, the IPitomy IP PBX can use VoIP SIP Trunks, replacing traditional PSTN lines with a broadband telephone service.

== Benefits of VoIP Technology ==

The IPitomy IP PBX can support any or all of these connectivity methods simultaneously or in any combination. Customers not quite ready to depend on VoIP providers for all of their business communications can start at their own pace and gain a comfort level, shifting to VoIP broadband providers at their own pace.

Benefits of VoIP technology include:

*'''One Wiring System:''' The system uses a single wiring system for telephones and dataall data and voice are on Local Area Network (LAN) Category 5 wiring.
*Software Based Upgrades: The IPitomy IP-PBX does not require any additional hardware to add extensions(phones) or advanced software features; ACD, Alerts, Scheduled Announcements, etc.
*'''Web-based Administration''': System administration is performed on the network through a Web-based administration program. The Web-Based Administration can be used locally or remotely from anywhere.
*'''Remote Users''': When calls are routed over the Internet, long distance charges can be avoided. In businesses with remote workers, these employees can stay logged into the office through a broadband connection at all times without incurring any additional charges. Remote users have all of the features of the local users. Remote users can be included in any ring groups, ACD (Automatic Call Distribution) Queues and other call routing schemes.
*'''Centralized System Features''': Every extension that is logged into the system is capable of receiving and originating calls. The use of system features such as voicemail, automated attendant and email are all centralized simplifying all support and maintenance.
*'''Reduced Costs''': VoIP system users can reduce cost in many areas of a business. VoIP telephony lowers the cost of support and maintenance costs, as well as, reducing telephony line costs by up to 50%.
*'''Simplifies Administration''': Moves, additions and changes are simple. The IPitomy IP PBX provides enhanced capabilities for users to make changes without incurring a service call.
*'''Investment Protection''': VoIP, and in particular, Session Initiation Protocol (SIP)-based VoIP products offer investment protection. The industry is rapidly moving toward Internet Protocol (IP) communications technologies. Older digital and analog technologies are becoming obsolete and are being replaced with IP-based products that will be around for a long time.

== IPitomy IP PBX Features ==

Understanding the IPitomy IP PBX’s architecture and how it works will make installing the system simple.

The IPitomy IP PBX is an all-in-one business communications system. This powerful system includes a complete suite of business communication applications in one appliance:

*Fully-featured Business Phone System
*Automated Attendant and Interactive Voice Response (IVR)
*Enhanced Call Distribution
*Enhanced Voice Messaging System with Unified Messaging
*Meet-me Conference Application
*Built-in Music on Hold
*Call Queuing for Inbound Calls
*Find Me/Follow Me
*Remote Extensions
*Browser-based Administration
*Branch Offices
*[http://wiki.ipitomy.com/wiki/ACD Automatic Call Distribution (ACD)]
*[http://wiki.ipitomy.com/wiki/Call_Recording Call Recording]
*[http://wiki.ipitomy.com/wiki/Routing_Ops Advanced Inbound Routing]
*[[Alerts]]
*[[IPPBX IMM Scheduled Calls|Scheduled Calling]]
*[[Room Management System|Hospitality PMS (Property Management Integration) Integration & Room Management]]
*[[Q Manager|QManager]]
*[[IPitomy Cloud Connect|Mobile]] & [[IPitomy Communicator|Desktop]] Softphones

The IPitomy IP PBX’s administration menus are a series of Web pages accessible from a Web browser. To the left of the Menu is a navigation bar that allows users to click on and administer each section of the system. Administration of the IPitomy IP PBX is simple and intuitive. The system is designed with seven primary areas of functionality and security:

*'''System''': System setup consists of network configuration settings.
*'''Providers''': Providers are sources of PSTN and VoIP connectivity. Providers are the lines that handle all incoming and outgoing calls. All VoIP and traditional telephone providers are setup here. DID numbers are also entered here.
*'''Destinations''': Destinations are places where calls are routed in the system: extensions, groups of extensions, automated attendants, conferences, and voicemail.
*'''Call Routing''': These settings route inbound calls to specific destinations within the system, and send outbound calls over specific local, long distance, international, and emergency routes.
*'''PBX Setup''': These settings globally configure PBX timers, voice messaging, and other system features.
*'''Reporting''': These reports display system usage, monitor activity, and provide diagnostic information.
*'''Diagnostics''': Additional diagnostic and testing options.
*'''Security Features''': Log, watch, and ban. Access Control Lists.

{| class="wikitable"
|-
| <center>Feature</center>
| <center>Description</center>
|-
| Extensions 
| Extensions are telephones. A telephone can be an IP (SIP) telephone or a Softphone. Calls are routed to an extension where people answer them. In the IPitomy IP PBX, an extension can be located in an office or outside the office where a broadband connection is used.
|-
| Groups 
| Groups are a set of extensions. Once a group is created, extensions can be designated as members of the group. This is accomplished by selecting group members from a drop- down list. Calls can be routed to groups via inbound routing.
|-
| Menus (Automated Attendant) 
|
To create an automated attendant use the system’s Menus feature. The Menus feature allows you to route calls to a destination in the system like a group, extension or another menu.

Call Destinations are selected from a drop-down list for each corresponding key-pad digit a caller must select to get to their chosen destination. A Menu must have a Menu Prompt. This is a recording that identifies for callers the destinations they may choose. For example, a Menu Prompt might offer callers the option to press “1” for Sales, “2” for Accounts Receivable or other digits for another department.

|-
| [http://wiki.ipitomy.com/wiki/Menu_Management Menu Management] 
| This feature allows the user to administer, update, and change the Menu (Auto Attendant) remotely using just a telephone (with DTMF dial capability).
|-
| Voicemail and Unified Messaging 
| When an extension is created, a voicemail box for that extension is also created. A voicemail box allows a caller to leave a message if a person is not available at the extension. When dialing into a mailbox for the first time, a user should record their name and a mailbox greeting. The name is used in the company’s dial-by-name directory when selected from the auto attendant (Menus). The greeting is played when they are not available to take a call and a caller reaches their mailbox.
If an email address is included in the Extension page, you can configure Unified Messaging and a copy of the voicemail message will be emailed as a .Wav file to the users email account. This message can then be listened to on a PC.

|-
| Directory 
| The system has a dial-by-name directory. This option may be part of the automated-attendant. When this option is selected, a caller dials the first three letters of the last / first name of the party they would like to reach. Names that match these three letters are played and the caller selects the extension to which they want to be transferred. Names are stated in the directory as they have been recorded by users in their voicemail box, and spelt out if they have not recorded their name.
|-
| Direct Inward Dialing (DID) Numbers 
| A Direct Inward Dialed (DID) number is a telephone number assigned by a service provider (i.e., T1 line, PRI or VoIP). DIDs allow direct routing of a call to a destination within the system. You can route to any destination available on the PBX.
|-
| Conferencing (Meet Me) 
| A Meet-me Conference is an extension on the system used for conference calls. Participants can access a conference by dialing the designated Meet-me Conference extension. Routing callers to a Meet-me Conference can be accomplished by using a DID, a menu, or simply transferring callers to the conference extension.
|-
| Follow-Me 
| This feature allows the PBX to try and find users who are not at their desk. It can be configured to call their cell phones, house phones, or other extensions in the PBX. Once answered, the user can accept the call, or refuse it. Unhandled calls return to the PBX to leave a message at the original extension’s voicemail.
|-
| Forwarding Gateway 
| Mobility has become a part of everyday life for most people. System users need to be able to take calls anywhere. The IPitomy IP PBX has the ability to forward calls. Users can turn call forwarding “on” and “off” while in the office or away from the office by using a touch-tone key pad. This is set up in the edit Extensions page, but can be modified from any phone, including a cell phone. Modifying forward settings remotely requires the automated attendant (Menus) option to be programmed.
|-
| Cascading Message Notification 
| This feature works using the same methods as FollowMe, but pertains to voicemail messages. When configured, if an extension gets a new voicemail, you will be able to send the voicemail message to a variety of numbers (Destinations), define the order in which to send the message and can be set to make the system to notify you that a new message was received. Additionally, you can add or remove extensions to the list of recipients when a broadcast message is sent.
|-
| Voicemail Gateway 
| Using either a Menu or a DID, users can call in from any telephone and check messages. The voicemail gateway allows users to dial a pre-defined digit from a touch- tone key pad on any phone to retrieve their messages.
|-
| Branch Offices 
| Branch offices can be created to allow multiple PBXs to route calls to each other. Branch office extensions can be transferred to, placed in ring groups, or selected as menu destinations.
|-
| Automatic Call Recording 
| A licensed feature that allows you to set up for automatic recording of calls, inbound via Ring Groups or outbound via Outbound Routes.
|-
| Advanced Inbound Routing 
| A licensed feature that allows for calls to be routed inbound based on a number of options, including inbound CID or digits entered at a prompt.
|-
|Alerts
|A licensed feature: That can be set up to allow notifications to be sent out when a certain number is dialed as an outgoing call, such as 911.
|-
|Scheduled Calling
|A licensed feature that allows for periodic automated calls to be made by the PBX. This can be used for announcements, bells, alarms etc.
|-
|Hospitality & Room Management
|A licensed feature which allows you to specify PBX extensions as hotel guest room extensions, and manage the rooms associated with them
|-
|QManager
|A licensed feature: Q Manager is a '''Windows application''' that allows users to monitor and interact with calls
|-
|Softphones
|A licensed feature: IPitomy offers mobile and desktop softphones that give you the ability to instantly deploy remote/work from home or new extensions
|}

''Table 1IP PBX Features''

IP PBX Manual Table of Contents

2022-09-06T13:24:21Z

Tyler:

__NOTITLE__ {{IP_PBX_Manual|sortkey=Table of Contents}}
<center>
'''IPitomy IP PBX Administrator Guide'''
</center><center>All materials in this documentation are proprietary and considered confidential to IPitomy Communications, LLC and may not be disclosed without the express written permission of IPitomy Communications, LLC. © 2012 IPitomy Communications, LLC All rights reserved.</center><center>'''IPitomy Communications, LLC'''</center><center>Phone: 941.306.2200</center><center>Email: [mailto:info@ipitomy.com info@ipitomy.com]</center><center>[http://www.ipitomy.com/ www.ipitomy.com]</center><center>Corporate Offices:
2837 Cattleman RD

Sarasota, FL, 34232
</center>
{| align="center" border="0" cellpadding="1" cellspacing="1"
|-
|
= '''Table of Contents''' =

*[[IPPBX IMM Intro|'''IPitomy's IP PBX (Intro)''']]
**[[IPPBX IMM Getting Started|'''Getting Started''']]
**[[IPPBX IMM DataNetworkConfig|'''Data Network Configuration''']]
*[[IPPBX IMM SystemAdmin|'''System Administration Intro''']]
*[[IPPBX IMM AdminSystemNetworking|'''System Networking''']]
*[[IP_PBX_Manual_System_Networking#UI_Users_.26_Groups|'''UI Users & Admin Groups''']]
*[[IPPBX IMM AdminProviders|'''Providers''']]
*[[IPPBX IMM SystemAdminDestinationsIntro|'''Destinations Intro''']]
**[[IP PBX Manual Extensions|Extensions]]
***[[IPPBX IMM SystemAdminDestinationsExtAddDel|Add Del]]
***[[IPPBX IMM SystemAdminDestinationsExtProvAutoDiscovery|Auto Discovery]]
***[[IPPBX IMM SystemAdminDestinationsExtEditPhoneSettings|Phone Configuration settings]]
**[[IPPBX IMM SystemAdminDestinationsGroups|Groups]]
**[[IPPBX IMM SystemAdminDestinationsMenus|Menus]]
**[[IPPBX IMM SystemAdminDestinationsConferences|Conference]]
**[[IPPBX IMM SystemAdminDestinationsVM|Voice Mail]]
**[[IPPBX IMM SystemAdminDestinationsSchedules|Schedule]]
**[[IPPBX IMM SystemAdminDestinationsBranchOffices|Branch Offices]]
**[[IPPBX IMM Scheduled Calls|Scheduled Calls]]
*Applications
**[[Call Recording]]
**[[Alerts]]
**[[Room Management System]]
*[[IPPBX IMM SystemAdminCallRouting|'''Call Routing''']]
*'''PBXSetup'''
**[[IPPBX_IMM_SystemAdminDestinationsGroups#Add_Automatic_Call_Distribution_.28ACD.29_Agents|Agents]]
**[[IPPBX IMM SystemAdminPBXSetupGeneralM|General]]
**[[IPPBX IMM SystemAdminPBXSetupDatabase|Database]]
**[[IPPBX IMM SystemAdminPBXSetupVM|Voicemail]]
**[[IPPBX IMM SystemAdminPBXSetupSIPSetup|SIP Setup (Global)]]
**[[IPPBX IMM SystemAdminPBXSetupPhone Global|Phone Global]]
**[[Mobile]]
**[[Chat]]
**[[IPPBX IMM SystemAdminPBXSetupPrompts|Prompts]]
**[[IPPBX IMM SystemAdminPBXSetupMOH|Music On Hold]]
**[[IPPBX IMM SystemAdminPBXSetupFeatureCodes|Feature Codes]]
**[[Screen Pop URL]]
**[[IPPBX IMM SystemAdminPBXSetupServices|Services]]
**[[IPPBX IMM SystemAdminReporting|Reporting]]
**[[IPPBX IMM SystemAdminReporting|Reports]]
**[[IPPBX IMM SystemAdminDiagnostics|Diagnostics]]
**[[IPPBX IMM SystemAdminMonitoring|Monitoring]]
**[[Queue Monitoring]]
*Web Applications
**[[Smart Personal Console]]
*Desktop Applications
**[[Desktop Call Manager|Desktop Call Manager]]
**[[Q_Manager|Queue Manager]]
*Diagnostics
**[[Network]]
**[[Packet Capture]]
*Appendicies
**[[IPPBX IMM SystemAdminAppendix1KeyTypesCodes|Appendix 1 Key Types, Codes]]
**[[HD Phones|Appendix 2 HD Phones]]
**[[IPPBX IMM SystemAdminAppendix3SoftPhones|Appendix 3 Soft Phones]]
**[[IPPBX IMM SystemAdminAppendix4IPAddresses|Appendix 4 IP Addresses]]
**[[IPPBX IMM SystemAdminAppendix5DHCPSettings|Appendix 5 DHCP Settings]]
**[[IPPBX IMM SystemAdminAppendix6RouterConfiguration|Appendix 6 Router Configuration]]
**[[IPPBX IMM SystemAdminAppendix7NetworkConsole|Appendix 7 Network Console]]
**[[IPPBX IMM SystemAdminAppendix8SoftwareUpgrade|Appendix 8 Software Upgrade]]
**[[IPPBX IMM SystemAdminAppendix9AastraPhones|Appendix 9 Aastra Phones]]
**[[IPPBX IMM SystemAdminAppendix10TroubleShooting|Appendix 10 Troubleshooting]]
*[[IPPBX IMM Glossary|Glossary]]

|}