IPitomy Wiki - User contributions [en]

File:Speaker diagram.pdf

2017-08-24T14:59:56Z

Nick Branica: Installation diagram for IP760 SIP Speakers -> Creation failed: Systemcheck showed missing or incorrect configurated, dependent commands!

Installation diagram for IP760 SIP Speakers

Training:Trunks

2017-08-21T17:26:15Z

Nick Branica: /* PRI */

= Training – Basic Trunking =

 

== Trunks ==

 

Trunks refers to the interconnection between your PBX and the provider to which the PBX is connected for PSTN interface.

 

There are two types of trunk interfaces:

1.      Analog

2.      Digital

There are two types of Digital interfaces:

1.      PRI/T-1

2.      SIP

 

When considering Digital connections; PRI requires a hardware board be installed into the IP PBX chassis. SIP does not.

When using analog lines (quasi “trunks”); hardware is always required.

 

== Analog Lines (loosely referred to as Analog Trunks) ==

Analog is a necessary evil in telephony today due to 100+ years of deployed analog circuits. The service provider of analog circuits is most commonly referred to as the Central Office or CO. A Central Office is really an Exchange. By comparison the IPitomy PBX is a "Private Branch Exchange" (PBX). Maybe you can see already that the trunking connectivity best suited to facilitate the Private Branch eXchange will be that connection most similar to the connection between inter-CO eXchanges: digital TRUNKS.

 

'''Analog Lines''' from the CO are the point of entry to the PSTN for many small businesses. An analog line is most similar to the line used to provide service to any telephone company subscriber. In the pictures notice that the lines are terminated in a very similar fashion regardless of their termination location (house or business). These lines are usually referred to as POTS (Plain Old Telephone Service) lines.

[[File:COtoHouseSmall.png]] [[File:COtoSmallBusinessSmall.png]]

For many years small businesses have deployed telephone systems called Key Systems which were used to distribute the analog lines onto each desktop for the people of the small business to interact more easily with callers. This solution in some ways has become a crutch for businesses to squeeze all that they can out of their line subscription. It is however inefficient due to the limitation of the "line" circuit, signal loss and limited feature capability and this approach is increasingly problematic. 

 

Analog lines in their original form were extended from the Central Office to the customer point of presence on copper wires. Today these copper wires are over-utilized by pair-gain devices used to extend and expand call capacity over the same wires. The result is multiple conversions of analog to an early digital signal using concentrators and multiplexors. The result is the delivery of analog service over quasi "copper" media which is at best an elaborate Rube Goldberg drawing.

[[File:SubscriberLoopSmall.gif]] 

IPitomy cannot in good conscience recommend analog lines. Analog lines are usually only required when the end user application requires very few connections to the CO and/or they are unwillingly to invest in better communications for their business.

 

When analog lines are used there two ways to interface them into the IP platform:

#Hardware interface board – classified PSTN Interfaces
#PSTN Gateways

In either case the need of proper CO line conditions is paramount in achieving acceptable results. We found a very good website for preparing for connection to the PSTN using '''CO Lines''' at: [http://www.sandman.com/ www.sandman.com]

Use the chart on this site at: [http://www.sandman.com/loop.html http://www.sandman.com/loop.html] to assure that your CO lines are within the required specifications.

We know that PSTN Gateways and PSTN interface boards only perform optimally when the CO Lines are at the '''desired '''levels.

== Analog Line Ongoing Maintenance and Testing  ==

IPitomy wants to make sure that you are aware, when you deploy analog "trunks" be prepared for ongoing maintenance and testing. Be ready to fill out the Sandman chart again-and-again.

[[File:SandmanLineTestSmall.gif]]

 

== Digital Trunks ==

When using digital trunks; hardware is only necessary when using PRI / T-1 circuits.

Digital trunks provide a higher level of connectivity between your POP (Point of Presence) and the CO or Digital-SIP Provider. Since the connection is digital it allows interaction between the connected eXchanges that control call connection and terminations as well as other features. A hugely beneficial feature of digital trunking is DID (Direct Inward Dialing). In addition to utilizing digital channels rather than lines, digital trunks allow signaling between the digital eXchanges such that call control can be passed from one to the other. DID is the means of selecting a specific user served by an eXchange from outside that eXchange - simply stated, DID allows PSTN callers to call a specific user or eXchange (IPitomy) destination by the number that they dial over the same service that serves all other users of the trunk.

 

=== PRI ===

 

IPitomy recommends the use of SIP as the “go-to” interface to the PSTN. If your customer requires use of a PRI, a PRI option is available . PRI circuits have the added advantage of being a guaranteed service – this means that although the bandwidth is limited (by todays norm) to 1.54Mbps, it will be guaranteed at that level from the service provider. This is not the case with other bandwidth service providers.

PRI in North America is 23 - "B" (Bearer) channels and 1 - "D" (Delta [data]) channel.

NOTE: When connecting a T1/PRI to the PBX, it is advised to use individually shielded pair T1 Cabling, not Cat-5 or Cat-6. Here is a great link explaining exactly why you want to avoid category cabling.

http://www.quabbin.com/page2027.html

[[File:Isdnpri.gif]]

PRI Advantages are:

*Digital connection
*Industry Proven technology and performance
*DID available at very competitive rates
*Guaranteed service delivery
*Some providers allow custom Caller ID feature functions
*23 talk paths (inbound and outbound use)

PRI Disadvantages:

*Traditional telephony costs (which have a cost – break-even point compared to their undesirable PSTN counterpart; POTS lines)
*Some providers do not allow use of some of the features that are available to an IP PBX user. Most noted restriction is:
**Custom Caller ID on outbound calls
**Providers can be slow to react to network outages

 

=== SIP Trunks ===

When you consider the use of SIP trunks, we want you to know that IPitomy fully embraces SIP technology. IPitomy is built on this technology which has become the most widely adopted VoIP protocol in the industry. There are a multitude of CODEC algorithms available with the click of your mouse giving you the range of voice-band constrained G.729 for bandwidth optimization to original telephone-grade call quality connections possible using the G.711 CODEC.

It is important to understand that SIP on the LAN is much different than SIP trunks.

The conditions of the LAN can be controlled and optimized. The LAN is where the IPitomy PBX and telephones reside. This makes utilization of SIP the best option for all PBX communications functions.

SIP trunks on the other hand are subject to the conditions of the WAN which are out of IPitomy’s and your control.

That said, SIP Trunks are digital and usually allow for all features available in IP.

 

The Advantages of SIP trunks are:

*Digital Connection
*Most competitive connection to telephony services and PSTN
*DID available – often at competitive rates
*Custom Caller ID possible in most cases
*No hardware required – SIP is a software only solution

SIP Disadvantages:

*Service delivery not guaranteed
*Call quality subject to fluctuating conditions on the WAN and shared bandwidth usage demand peaks
**Represent this solution as that comparable to the service one might expect while using a cell phone – some calls are excellent, others are not.
*Requires you to be prepared in terms of broadband services available at the point of presence.
**Test broadband connection for true-to-life conditions that might be expected when SIP trunks are deployed.

 

== FAX ==

FAX is a technology based on "modem" communication. As such, it was designed to be used with analog lines that were a copper pair extending from the Central Office to a business or residence. In today's world, the direct copper connection is more often than not replaced with a shared digital connection that is not exactly the same as the copper connection in many cases. . It was great when developed – FAX over the Internet has replaced the old fashioned copper connection as being the most reliable of FAX delivery methods.

There are three types of FAX that can be used with SIP systems:

1.) FAX over the Internet. Using a traditional FAX machine, a box is installed with an analog port to connect to the FAX machine like a phone line. A LAN connection is included so the FAX can be sent using a reliable transmission method to assure that it arrives at the destination without any errors. Inbound FAX works the same with highly reliable servers that send the Document to the FAX machine without errors. When the customer prefers to implement fax using a FAX machine, this is the preferred method. This service also includes the digital FAX benefits listed below. IPitomy sells this service and can be purchased directly from IPitomy.

2.) Digital FAX. Digital FAX allows incoming FAX to arrive as an email in an Email inbox. It is a reliable FAX over Internet process. Outbound Fax documents can be sent from the desktop using a driver that installs on the user PC that acts similar to a printer driver. When a customer is comfortable using a scanner of created documents for outbound faxes, this is a highly reliable method and is less expensive than having the FAX over Internet device plugged into the FAX machine. This FAX Service can be purchased through IPitomy.

3.) Traditional copper lines or use of an analog gateway. This is best if using an analog line, but will not provide the excellent results as the two methods above.

Training:Trunks

2017-08-21T15:45:01Z

Nick Branica: /* FAX */

= Training – Basic Trunking =

 

== Trunks ==

 

Trunks refers to the interconnection between your PBX and the provider to which the PBX is connected for PSTN interface.

 

There are two types of trunk interfaces:

1.      Analog

2.      Digital

There are two types of Digital interfaces:

1.      PRI/T-1

2.      SIP

 

When considering Digital connections; PRI requires a hardware board be installed into the IP PBX chassis. SIP does not.

When using analog lines (quasi “trunks”); hardware is always required.

 

== Analog Lines (loosely referred to as Analog Trunks) ==

Analog is a necessary evil in telephony today due to 100+ years of deployed analog circuits. The service provider of analog circuits is most commonly referred to as the Central Office or CO. A Central Office is really an Exchange. By comparison the IPitomy PBX is a "Private Branch Exchange" (PBX). Maybe you can see already that the trunking connectivity best suited to facilitate the Private Branch eXchange will be that connection most similar to the connection between inter-CO eXchanges: digital TRUNKS.

 

'''Analog Lines''' from the CO are the point of entry to the PSTN for many small businesses. An analog line is most similar to the line used to provide service to any telephone company subscriber. In the pictures notice that the lines are terminated in a very similar fashion regardless of their termination location (house or business). These lines are usually referred to as POTS (Plain Old Telephone Service) lines.

[[File:COtoHouseSmall.png]] [[File:COtoSmallBusinessSmall.png]]

For many years small businesses have deployed telephone systems called Key Systems which were used to distribute the analog lines onto each desktop for the people of the small business to interact more easily with callers. This solution in some ways has become a crutch for businesses to squeeze all that they can out of their line subscription. It is however inefficient due to the limitation of the "line" circuit, signal loss and limited feature capability and this approach is increasingly problematic. 

 

Analog lines in their original form were extended from the Central Office to the customer point of presence on copper wires. Today these copper wires are over-utilized by pair-gain devices used to extend and expand call capacity over the same wires. The result is multiple conversions of analog to an early digital signal using concentrators and multiplexors. The result is the delivery of analog service over quasi "copper" media which is at best an elaborate Rube Goldberg drawing.

[[File:SubscriberLoopSmall.gif]] 

IPitomy cannot in good conscience recommend analog lines. Analog lines are usually only required when the end user application requires very few connections to the CO and/or they are unwillingly to invest in better communications for their business.

 

When analog lines are used there two ways to interface them into the IP platform:

#Hardware interface board – classified PSTN Interfaces
#PSTN Gateways

In either case the need of proper CO line conditions is paramount in achieving acceptable results. We found a very good website for preparing for connection to the PSTN using '''CO Lines''' at: [http://www.sandman.com/ www.sandman.com]

Use the chart on this site at: [http://www.sandman.com/loop.html http://www.sandman.com/loop.html] to assure that your CO lines are within the required specifications.

We know that PSTN Gateways and PSTN interface boards only perform optimally when the CO Lines are at the '''desired '''levels.

== Analog Line Ongoing Maintenance and Testing  ==

IPitomy wants to make sure that you are aware, when you deploy analog "trunks" be prepared for ongoing maintenance and testing. Be ready to fill out the Sandman chart again-and-again.

[[File:SandmanLineTestSmall.gif]]

 

== Digital Trunks ==

When using digital trunks; hardware is only necessary when using PRI / T-1 circuits.

Digital trunks provide a higher level of connectivity between your POP (Point of Presence) and the CO or Digital-SIP Provider. Since the connection is digital it allows interaction between the connected eXchanges that control call connection and terminations as well as other features. A hugely beneficial feature of digital trunking is DID (Direct Inward Dialing). In addition to utilizing digital channels rather than lines, digital trunks allow signaling between the digital eXchanges such that call control can be passed from one to the other. DID is the means of selecting a specific user served by an eXchange from outside that eXchange - simply stated, DID allows PSTN callers to call a specific user or eXchange (IPitomy) destination by the number that they dial over the same service that serves all other users of the trunk.

 

=== PRI ===

 

IPitomy recommends the use of PRI as the “go-to” interface to the PSTN. If your customer demands the best performance with high-functionality, PRI is the only choice. PRI circuits have the added advantage of being a guaranteed service – this means that although the bandwidth is limited (by todays norm) to 1.54Mbps, it will be guaranteed at that level from the service provider. This is not the case with other bandwidth service providers.

PRI in North America is 23 - "B" (Bearer) channels and 1 - "D" (Delta [data]) channel.

NOTE: When connecting a T1/PRI to the PBX, it is advised to use individually shielded pair T1 Cabling, not Cat-5 or Cat-6. Here is a great link explaining exactly why you want to avoid category cabling.

http://www.quabbin.com/page2027.html

[[File:Isdnpri.gif]]

PRI Advantages are:

*Digital connection
*Industry Proven technology and performance
*DID available at very competitive rates
*Guaranteed service delivery
*Some providers allow custom Caller ID feature functions
*23 talk paths (inbound and outbound use)

PRI Disadvantages:

*Traditional telephony costs (which have a cost – break-even point compared to their undesirable PSTN counterpart; POTS lines)
*Some providers do not allow use of some of the features that are available to an IP PBX user. Most noted restriction is:
**Custom Caller ID on outbound calls

 

=== SIP Trunks ===

When you consider the use of SIP trunks, we want you to know that IPitomy fully embraces SIP technology. IPitomy is built on this technology which has become the most widely adopted VoIP protocol in the industry. There are a multitude of CODEC algorithms available with the click of your mouse giving you the range of voice-band constrained G.729 for bandwidth optimization to original telephone-grade call quality connections possible using the G.711 CODEC.

It is important to understand that SIP on the LAN is much different than SIP trunks.

The conditions of the LAN can be controlled and optimized. The LAN is where the IPitomy PBX and telephones reside. This makes utilization of SIP the best option for all PBX communications functions.

SIP trunks on the other hand are subject to the conditions of the WAN which are out of IPitomy’s and your control.

That said, SIP Trunks are digital and usually allow for all features available in IP.

 

The Advantages of SIP trunks are:

*Digital Connection
*Most competitive connection to telephony services and PSTN
*DID available – often at competitive rates
*Custom Caller ID possible in most cases
*No hardware required – SIP is a software only solution

SIP Disadvantages:

*Service delivery not guaranteed
*Call quality subject to fluctuating conditions on the WAN and shared bandwidth usage demand peaks
**Represent this solution as that comparable to the service one might expect while using a cell phone – some calls are excellent, others are not.
*Requires you to be prepared in terms of broadband services available at the point of presence.
**Test broadband connection for true-to-life conditions that might be expected when SIP trunks are deployed.

 

== FAX ==

FAX is a technology based on "modem" communication. As such, it was designed to be used with analog lines that were a copper pair extending from the Central Office to a business or residence. In today's world, the direct copper connection is more often than not replaced with a shared digital connection that is not exactly the same as the copper connection in many cases. . It was great when developed – FAX over the Internet has replaced the old fashioned copper connection as being the most reliable of FAX delivery methods.

There are three types of FAX that can be used with SIP systems:

1.) FAX over the Internet. Using a traditional FAX machine, a box is installed with an analog port to connect to the FAX machine like a phone line. A LAN connection is included so the FAX can be sent using a reliable transmission method to assure that it arrives at the destination without any errors. Inbound FAX works the same with highly reliable servers that send the Document to the FAX machine without errors. When the customer prefers to implement fax using a FAX machine, this is the preferred method. This service also includes the digital FAX benefits listed below. IPitomy sells this service and can be purchased directly from IPitomy.

2.) Digital FAX. Digital FAX allows incoming FAX to arrive as an email in an Email inbox. It is a reliable FAX over Internet process. Outbound Fax documents can be sent from the desktop using a driver that installs on the user PC that acts similar to a printer driver. When a customer is comfortable using a scanner of created documents for outbound faxes, this is a highly reliable method and is less expensive than having the FAX over Internet device plugged into the FAX machine. This FAX Service can be purchased through IPitomy.

3.) Traditional copper lines or use of an analog gateway. This is best if using an analog line, but will not provide the excellent results as the two methods above.

Training:Trunks

2017-08-21T15:25:58Z

Nick Branica: /* FAX */

= Training – Basic Trunking =

 

== Trunks ==

 

Trunks refers to the interconnection between your PBX and the provider to which the PBX is connected for PSTN interface.

 

There are two types of trunk interfaces:

1.      Analog

2.      Digital

There are two types of Digital interfaces:

1.      PRI/T-1

2.      SIP

 

When considering Digital connections; PRI requires a hardware board be installed into the IP PBX chassis. SIP does not.

When using analog lines (quasi “trunks”); hardware is always required.

 

== Analog Lines (loosely referred to as Analog Trunks) ==

Analog is a necessary evil in telephony today due to 100+ years of deployed analog circuits. The service provider of analog circuits is most commonly referred to as the Central Office or CO. A Central Office is really an Exchange. By comparison the IPitomy PBX is a "Private Branch Exchange" (PBX). Maybe you can see already that the trunking connectivity best suited to facilitate the Private Branch eXchange will be that connection most similar to the connection between inter-CO eXchanges: digital TRUNKS.

 

'''Analog Lines''' from the CO are the point of entry to the PSTN for many small businesses. An analog line is most similar to the line used to provide service to any telephone company subscriber. In the pictures notice that the lines are terminated in a very similar fashion regardless of their termination location (house or business). These lines are usually referred to as POTS (Plain Old Telephone Service) lines.

[[File:COtoHouseSmall.png]] [[File:COtoSmallBusinessSmall.png]]

For many years small businesses have deployed telephone systems called Key Systems which were used to distribute the analog lines onto each desktop for the people of the small business to interact more easily with callers. This solution in some ways has become a crutch for businesses to squeeze all that they can out of their line subscription. It is however inefficient due to the limitation of the "line" circuit, signal loss and limited feature capability and this approach is increasingly problematic. 

 

Analog lines in their original form were extended from the Central Office to the customer point of presence on copper wires. Today these copper wires are over-utilized by pair-gain devices used to extend and expand call capacity over the same wires. The result is multiple conversions of analog to an early digital signal using concentrators and multiplexors. The result is the delivery of analog service over quasi "copper" media which is at best an elaborate Rube Goldberg drawing.

[[File:SubscriberLoopSmall.gif]] 

IPitomy cannot in good conscience recommend analog lines. Analog lines are usually only required when the end user application requires very few connections to the CO and/or they are unwillingly to invest in better communications for their business.

 

When analog lines are used there two ways to interface them into the IP platform:

#Hardware interface board – classified PSTN Interfaces
#PSTN Gateways

In either case the need of proper CO line conditions is paramount in achieving acceptable results. We found a very good website for preparing for connection to the PSTN using '''CO Lines''' at: [http://www.sandman.com/ www.sandman.com]

Use the chart on this site at: [http://www.sandman.com/loop.html http://www.sandman.com/loop.html] to assure that your CO lines are within the required specifications.

We know that PSTN Gateways and PSTN interface boards only perform optimally when the CO Lines are at the '''desired '''levels.

== Analog Line Ongoing Maintenance and Testing  ==

IPitomy wants to make sure that you are aware, when you deploy analog "trunks" be prepared for ongoing maintenance and testing. Be ready to fill out the Sandman chart again-and-again.

[[File:SandmanLineTestSmall.gif]]

 

== Digital Trunks ==

When using digital trunks; hardware is only necessary when using PRI / T-1 circuits.

Digital trunks provide a higher level of connectivity between your POP (Point of Presence) and the CO or Digital-SIP Provider. Since the connection is digital it allows interaction between the connected eXchanges that control call connection and terminations as well as other features. A hugely beneficial feature of digital trunking is DID (Direct Inward Dialing). In addition to utilizing digital channels rather than lines, digital trunks allow signaling between the digital eXchanges such that call control can be passed from one to the other. DID is the means of selecting a specific user served by an eXchange from outside that eXchange - simply stated, DID allows PSTN callers to call a specific user or eXchange (IPitomy) destination by the number that they dial over the same service that serves all other users of the trunk.

 

=== PRI ===

 

IPitomy recommends the use of PRI as the “go-to” interface to the PSTN. If your customer demands the best performance with high-functionality, PRI is the only choice. PRI circuits have the added advantage of being a guaranteed service – this means that although the bandwidth is limited (by todays norm) to 1.54Mbps, it will be guaranteed at that level from the service provider. This is not the case with other bandwidth service providers.

PRI in North America is 23 - "B" (Bearer) channels and 1 - "D" (Delta [data]) channel.

NOTE: When connecting a T1/PRI to the PBX, it is advised to use individually shielded pair T1 Cabling, not Cat-5 or Cat-6. Here is a great link explaining exactly why you want to avoid category cabling.

http://www.quabbin.com/page2027.html

[[File:Isdnpri.gif]]

PRI Advantages are:

*Digital connection
*Industry Proven technology and performance
*DID available at very competitive rates
*Guaranteed service delivery
*Some providers allow custom Caller ID feature functions
*23 talk paths (inbound and outbound use)

PRI Disadvantages:

*Traditional telephony costs (which have a cost – break-even point compared to their undesirable PSTN counterpart; POTS lines)
*Some providers do not allow use of some of the features that are available to an IP PBX user. Most noted restriction is:
**Custom Caller ID on outbound calls

 

=== SIP Trunks ===

When you consider the use of SIP trunks, we want you to know that IPitomy fully embraces SIP technology. IPitomy is built on this technology which has become the most widely adopted VoIP protocol in the industry. There are a multitude of CODEC algorithms available with the click of your mouse giving you the range of voice-band constrained G.729 for bandwidth optimization to original telephone-grade call quality connections possible using the G.711 CODEC.

It is important to understand that SIP on the LAN is much different than SIP trunks.

The conditions of the LAN can be controlled and optimized. The LAN is where the IPitomy PBX and telephones reside. This makes utilization of SIP the best option for all PBX communications functions.

SIP trunks on the other hand are subject to the conditions of the WAN which are out of IPitomy’s and your control.

That said, SIP Trunks are digital and usually allow for all features available in IP.

 

The Advantages of SIP trunks are:

*Digital Connection
*Most competitive connection to telephony services and PSTN
*DID available – often at competitive rates
*Custom Caller ID possible in most cases
*No hardware required – SIP is a software only solution

SIP Disadvantages:

*Service delivery not guaranteed
*Call quality subject to fluctuating conditions on the WAN and shared bandwidth usage demand peaks
**Represent this solution as that comparable to the service one might expect while using a cell phone – some calls are excellent, others are not.
*Requires you to be prepared in terms of broadband services available at the point of presence.
**Test broadband connection for true-to-life conditions that might be expected when SIP trunks are deployed.

 

== FAX ==

FAX is a technology based on "modem" communication. As such, it was designed to be used with analog lines that were a copper pair extending from the Central Office to a business or residence. In today's world, the direct copper connection is more often than not replaced with a shared digital connection that is not exactly the same as the copper connection in many cases. . It was great when developed – FAX over the Internet has replaced the old fashioned copper connection as being the most reliable of FAX delivery methods.

There are three types of FAX that can be used with SIP systems:

1.) FAX over the Internet. Using a traditional FAX machine, a box is installed with an analog port to connect to the FAX machine like a phone line. A LAN connection is included so the FAX can be sent using a reliable transmission method to assure that it arrives at the destination without any errors. Inbound FAX works the same with highly reliable servers that send the FAX to the FAX machine without errors. When the customer prefers to implement fax using a FAX machine, this is the preferred method. This service also includes the digital FAX benefits listed below. IPitomy sells this service and can be purchased directly from IPitomy.

2.) Digital FAX. Digital FAX allows incoming FAX to arrive as an email in an Email inbox. It is a reliable FAX over Internet process. Outbound Fax documents can be sent from the desktop using a driver that installs on the user PC that acts similar to a printer driver. When a customer is comfortable using a scanner of created documents for outbound faxes, this is a highly reliable method and is less expensive than having the FAX over Internet device plugged into the FAX machine. This FAX Service can be purchased through IPitomy.

3.) Traditional copper lines or use of an analog gateway. This is best if using an analog line, but will not provide the excellent results as the two methods above.

Training:Trunks

2017-08-21T15:16:50Z

Nick Branica: /* FAX */

= Training – Basic Trunking =

 

== Trunks ==

 

Trunks refers to the interconnection between your PBX and the provider to which the PBX is connected for PSTN interface.

 

There are two types of trunk interfaces:

1.      Analog

2.      Digital

There are two types of Digital interfaces:

1.      PRI/T-1

2.      SIP

 

When considering Digital connections; PRI requires a hardware board be installed into the IP PBX chassis. SIP does not.

When using analog lines (quasi “trunks”); hardware is always required.

 

== Analog Lines (loosely referred to as Analog Trunks) ==

Analog is a necessary evil in telephony today due to 100+ years of deployed analog circuits. The service provider of analog circuits is most commonly referred to as the Central Office or CO. A Central Office is really an Exchange. By comparison the IPitomy PBX is a "Private Branch Exchange" (PBX). Maybe you can see already that the trunking connectivity best suited to facilitate the Private Branch eXchange will be that connection most similar to the connection between inter-CO eXchanges: digital TRUNKS.

 

'''Analog Lines''' from the CO are the point of entry to the PSTN for many small businesses. An analog line is most similar to the line used to provide service to any telephone company subscriber. In the pictures notice that the lines are terminated in a very similar fashion regardless of their termination location (house or business). These lines are usually referred to as POTS (Plain Old Telephone Service) lines.

[[File:COtoHouseSmall.png]] [[File:COtoSmallBusinessSmall.png]]

For many years small businesses have deployed telephone systems called Key Systems which were used to distribute the analog lines onto each desktop for the people of the small business to interact more easily with callers. This solution in some ways has become a crutch for businesses to squeeze all that they can out of their line subscription. It is however inefficient due to the limitation of the "line" circuit, signal loss and limited feature capability and this approach is increasingly problematic. 

 

Analog lines in their original form were extended from the Central Office to the customer point of presence on copper wires. Today these copper wires are over-utilized by pair-gain devices used to extend and expand call capacity over the same wires. The result is multiple conversions of analog to an early digital signal using concentrators and multiplexors. The result is the delivery of analog service over quasi "copper" media which is at best an elaborate Rube Goldberg drawing.

[[File:SubscriberLoopSmall.gif]] 

IPitomy cannot in good conscience recommend analog lines. Analog lines are usually only required when the end user application requires very few connections to the CO and/or they are unwillingly to invest in better communications for their business.

 

When analog lines are used there two ways to interface them into the IP platform:

#Hardware interface board – classified PSTN Interfaces
#PSTN Gateways

In either case the need of proper CO line conditions is paramount in achieving acceptable results. We found a very good website for preparing for connection to the PSTN using '''CO Lines''' at: [http://www.sandman.com/ www.sandman.com]

Use the chart on this site at: [http://www.sandman.com/loop.html http://www.sandman.com/loop.html] to assure that your CO lines are within the required specifications.

We know that PSTN Gateways and PSTN interface boards only perform optimally when the CO Lines are at the '''desired '''levels.

== Analog Line Ongoing Maintenance and Testing  ==

IPitomy wants to make sure that you are aware, when you deploy analog "trunks" be prepared for ongoing maintenance and testing. Be ready to fill out the Sandman chart again-and-again.

[[File:SandmanLineTestSmall.gif]]

 

== Digital Trunks ==

When using digital trunks; hardware is only necessary when using PRI / T-1 circuits.

Digital trunks provide a higher level of connectivity between your POP (Point of Presence) and the CO or Digital-SIP Provider. Since the connection is digital it allows interaction between the connected eXchanges that control call connection and terminations as well as other features. A hugely beneficial feature of digital trunking is DID (Direct Inward Dialing). In addition to utilizing digital channels rather than lines, digital trunks allow signaling between the digital eXchanges such that call control can be passed from one to the other. DID is the means of selecting a specific user served by an eXchange from outside that eXchange - simply stated, DID allows PSTN callers to call a specific user or eXchange (IPitomy) destination by the number that they dial over the same service that serves all other users of the trunk.

 

=== PRI ===

 

IPitomy recommends the use of PRI as the “go-to” interface to the PSTN. If your customer demands the best performance with high-functionality, PRI is the only choice. PRI circuits have the added advantage of being a guaranteed service – this means that although the bandwidth is limited (by todays norm) to 1.54Mbps, it will be guaranteed at that level from the service provider. This is not the case with other bandwidth service providers.

PRI in North America is 23 - "B" (Bearer) channels and 1 - "D" (Delta [data]) channel.

NOTE: When connecting a T1/PRI to the PBX, it is advised to use individually shielded pair T1 Cabling, not Cat-5 or Cat-6. Here is a great link explaining exactly why you want to avoid category cabling.

http://www.quabbin.com/page2027.html

[[File:Isdnpri.gif]]

PRI Advantages are:

*Digital connection
*Industry Proven technology and performance
*DID available at very competitive rates
*Guaranteed service delivery
*Some providers allow custom Caller ID feature functions
*23 talk paths (inbound and outbound use)

PRI Disadvantages:

*Traditional telephony costs (which have a cost – break-even point compared to their undesirable PSTN counterpart; POTS lines)
*Some providers do not allow use of some of the features that are available to an IP PBX user. Most noted restriction is:
**Custom Caller ID on outbound calls

 

=== SIP Trunks ===

When you consider the use of SIP trunks, we want you to know that IPitomy fully embraces SIP technology. IPitomy is built on this technology which has become the most widely adopted VoIP protocol in the industry. There are a multitude of CODEC algorithms available with the click of your mouse giving you the range of voice-band constrained G.729 for bandwidth optimization to original telephone-grade call quality connections possible using the G.711 CODEC.

It is important to understand that SIP on the LAN is much different than SIP trunks.

The conditions of the LAN can be controlled and optimized. The LAN is where the IPitomy PBX and telephones reside. This makes utilization of SIP the best option for all PBX communications functions.

SIP trunks on the other hand are subject to the conditions of the WAN which are out of IPitomy’s and your control.

That said, SIP Trunks are digital and usually allow for all features available in IP.

 

The Advantages of SIP trunks are:

*Digital Connection
*Most competitive connection to telephony services and PSTN
*DID available – often at competitive rates
*Custom Caller ID possible in most cases
*No hardware required – SIP is a software only solution

SIP Disadvantages:

*Service delivery not guaranteed
*Call quality subject to fluctuating conditions on the WAN and shared bandwidth usage demand peaks
**Represent this solution as that comparable to the service one might expect while using a cell phone – some calls are excellent, others are not.
*Requires you to be prepared in terms of broadband services available at the point of presence.
**Test broadband connection for true-to-life conditions that might be expected when SIP trunks are deployed.

 

== FAX ==

FAX is a technology based on "modem" communication. As such, it was designed to be used with analog lines that were a copper pair extending from the Central Office to a business or residence. In today's world, the direct copper connection is more often than not replaced with a shared digital connection that is not exactly the same as the copper connection in many cases. . It was great when developed – FAX over the Internet has replaced the old fashioned copper connection as being the most reliable of FAX delivery methods.

There are three types of FAX that can be used with SIP systems:

1.) FAX over the Internet. Using a traditional FAX machine, a box is installed with an analog port to connect to the FAX machine like a phone line. A LAN connection is included so the FAX can be sent using a reliable transmission method to assure that it arrives at the destination without any errors. Inbound FAX works the same with highly reliable servers that send the FAX to the FAX machine without errors. When the customer prefers to implement fax using a FAX machine, this is the preferred method. This service also includes the digital FAX benefits listed below. IPitomy sells this service and can be purchased directly from IPitomy.

2.) Digital FAX. Digital FAX allows incoming FAX to arrive as an email in an Email inbox. It is a reliable FAX over Internet process. Outbound Fax documents can be sent from the desktop using a driver that installs on the user PC that acts similar to a printer driver. When a customer is comfortable using a scanner of created documents for outbound faxes, this is a highly reliable method and is less expensive than having the FAX over Internet device plugged into the FAX machine.

3.) Traditional copper lines or use of an analog gateway. This is best if using an analog line, but will not provide the excellent results as the two methods above.

Training:Site Survey

2017-08-15T17:49:48Z

Nick Branica: /* Trunks – what trunk types will be used? */

= IPitomy Technical Training – Basic =

== Site Survey ==

 

Performing a site survey is the best guard against surprises at cutover. The elements of an IP PBX involve both telephony and network readiness.

= Telephony Readiness: =

== Trunks – what trunk types will be used? ==

*IPitomy recommends the use of PRI trunks if reliable bandwidth is not available for SIP Trunking.
**Advantages are:
***High technical capabilities
***Have been regarded as highly reliable, but often have no redundancy
***Guaranteed level of service from the provider

*Quality SIP Trunks are recommended and may be used when network infrastructure at the WAN interface has adequate bandwidth.
**Advantages are:
***High technical capabilities, redundancy, High Quality
***Low Cost - More Capacity
***Easy to Configure
***Can be purchased direct from [[http://www.ipitomy.com/index.php/services/ipitomy-siptrunks IPitomy]]

*Analog trunks at times are a necessary evil. These are the least desirable trunk type. Build-in cost for additional testing and maintenance when you must use analog trunks. As part of the site survey you must know, for EACH Trunk circuit:
**On-Hook voltage DC (should be between 44 and 50VDC)
**Off-Hook voltage DC (should be between 6 and 9VDC)
**Off-Hook mA (loop current) (should be between 23 and 34 mA)
**On-Hook voltage (AC) (should be less than 5VAC
**Off-Hook voltage (AC)
**Off-Hook voltages (AC/DC) from each side (Tip/Ring) of the analog line to Earth Ground.
**Transient voltages – check for AC and DC voltage fluctuations on each lines while activities are present on the other lines (ringing on other lines, connected calls on other lines)
**Test Line Loss (measured in decibels) from ZERO. No loss is zero… typical line loss will be less than -3db (i.e. 0db to -3db)(FYI, Line loss can be as much as -9db before Telco’s are required to improve line conditions.)
**Read the Loop Current & Circuit Loss Technical Bulletin at  [http://www.sandman.com/loopcur.html#CircuitLoss http://www.sandman.com/loopcur.html#CircuitLoss]
**Line conditioning apparatus is available at Sandman.com. Use the link above to reach this useful site.
*Disadvantages of analog lines are:
**Low technical capabilities
**Mediocre 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

== Extensions – How many extensions and what models? ==

*Physically review each location where a telephone will be installed
*Become aware of the person that will use the telephone and grasp challenges that this person will encounter – will this user require additional training
*Prepare to train users – on-line training is available at IPitomy Documentation Repository: [https://skydrive.live.com/?cid=a31a942025b325ce&id=A31A942025B325CE%21104 https://skydrive.live.com/?cid=a31a942025b325ce&id=A31A942025B325CE%21104].

= Network Readiness =

The network is the ''Backplane'' of the IP PBX. In old telephony solutions (TDM, Switched Matrix) the backplane was designed and controlled by the PBX manufacturer – not so in IP Telephony. The network MUST be in pristine condition to allow use with voice. Voice traffic MUST 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 voice quality from these issues.

== Cabling – 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.

== WAN – ISP ==

*Who is the Internet Service Provider at the site?
*Should this change?
*Are there cost advantages that can be offered in association to ISP and Trunks (SIP or PRI)?
*What is the WAN connection bandwidth performance?<ul style="list-style-type:circle;">
<li>A good simple tester: [http://speedtest.net/ http://speedtest.net/]</li>
<li>Keep in mind that ISP bandwidth testers are momentary snap-shots of the performance of your WAN (Internet) connection.</li>
<li>Several tests should be performed during the hours of operation at the site where the IP PBX will be installed.</li>
<li>Network performance software is available for network monitoring over extended periods. (Google it)</li>
</ul></ul>

== Router ==

*What kind of Router is in use?
*Are there multiple Public IP Addresses<ul style="list-style-type:circle;">
<li>What is/are the Public IP Address(es)</li>
</ul><li>Is this router on the list of Reportedly Works Well routers on the IPitomy web site? [http://www.ipitomy.com/webrelease/IPitomy/Router%20Compatibility%20Guide.pdf http://www.ipitomy.com/webrelease/IPitomy/Router%20Compatibility%20Guide.pdf]</li>
<li>Can you use your own router in Gateway mode to allow all voice traffic to be routed through that router versus routing all data and voice traffic through the router on site?</li>
<li>Who is in charge of the router? (Person, IT Dept?)<ul style="list-style-type:circle;">
<li>If IT Dept. controls the router and/or Switches on the site you MUST become best of buddies with the group of people. They WILL impact the performance of the voice traffic on the network.</li>
</ul></li>
<li>What device will serve DHCP?</li>
<li>Can you take over control of the router<ul style="list-style-type:circle;">
<li>If so, replace it with one of those on the IPitomy list that you have tested and found to perform as you intend.</li>
</ul></li>
<li>Is the router capable of Port Forwarding?<ul style="list-style-type:circle;">
<li>The router will need this function. The more robust the better. It is desirable to set Port Forward rules to receive packets on any port designation given by the sending device and then to deliver these packets to any LAN IP Address with a unique Port assignment.</li>
</ul></li>
<li>What is the access (User Name & Password) of the router?</li>
<li>How many concurrent voice traffic sessions is this router capable of NAT’ing? (NAT is Network Address Translation.) This is essential to the ability of a router to handle Remote Telephones.</li>
<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 information.</li>
</ul></li>
<li>Is an ALG (Application Layer Gateway) present on the router?<ul style="list-style-type:circle;">
<li>Is the ALG enabled?</li>
<li>If so, this will need to be disabled or;</li>
<li>…complex router programming is likely required to assure that no voice traffic is impeded by the operation of the ALG
*ALG’s often contribute to one-way audio issues.
*ALG’s often inhibit Remote Phone operation</li>
</li>

== Sonicwall® ==

Is a Sonicwall® present? If so, significant programming will be required to assure that no voice traffic (BOTH LAN and WAN traffic) is impeded.

*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>
</ul></ul>

== 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:

*Are they managed (programmable)?
*Are there enough ports to accommodate the number of telephones that will be installed?
*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></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

= Worksheet =

Complete the IPitomy Setup Worksheet located at: [http://www.ipitomy.com/webrelease/IPitomy/IP1100+/ http://www.ipitomy.com/webrelease/IPitomy/IP1100+/] each time that an IP PBX is to be installed. This form will help to organize the deployment of the PBX by aiding you to gather the information needed to program and prepare the site application. The form also provides a hard-copy backup of the database configuration.

= 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: [http://www.ipitomy.com/webrelease/SIP%20Providers/ http://www.ipitomy.com/webrelease/SIP%20Providers/]. 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:

*Never assume that a network is performing at its highest efficiency
*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 usually are internet-based and are therefore a non-guaranteed voice connection service due to fluctuations in bandwidth and WAN network performance. (Where PRI is a guaranteed service from the provider.)</li>
</ul></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.

Training:Site Survey

2017-08-15T17:47:34Z

Nick Branica: /* Trunks – what trunk types will be used? */

= IPitomy Technical Training – Basic =

== Site Survey ==

 

Performing a site survey is the best guard against surprises at cutover. The elements of an IP PBX involve both telephony and network readiness.

= Telephony Readiness: =

== Trunks – what trunk types will be used? ==

*IPitomy recommends the use of PRI trunks if reliable bandwidth is not available for SIP Trunking.
**Advantages are:
***High technical capabilities
***Have been regarded as highly reliable, but often have no redundancy
***Guaranteed level of service from the provider

*Quality SIP Trunks may be used when network infrastructure at the WAN interface has adequate bandwidth.
**Advantages are:
***High technical capabilities, redundancy, High Quality
***Low Cost - More Capacity
***Easy to Configure
***Can be purchased direct from [[http://www.ipitomy.com/index.php/services/ipitomy-siptrunks IPitomy]]

*Analog trunks at times are a necessary evil. These are the least desirable trunk type. Build-in cost for additional testing and maintenance when you must use analog trunks. As part of the site survey you must know, for EACH Trunk circuit:
**On-Hook voltage DC (should be between 44 and 50VDC)
**Off-Hook voltage DC (should be between 6 and 9VDC)
**Off-Hook mA (loop current) (should be between 23 and 34 mA)
**On-Hook voltage (AC) (should be less than 5VAC
**Off-Hook voltage (AC)
**Off-Hook voltages (AC/DC) from each side (Tip/Ring) of the analog line to Earth Ground.
**Transient voltages – check for AC and DC voltage fluctuations on each lines while activities are present on the other lines (ringing on other lines, connected calls on other lines)
**Test Line Loss (measured in decibels) from ZERO. No loss is zero… typical line loss will be less than -3db (i.e. 0db to -3db)(FYI, Line loss can be as much as -9db before Telco’s are required to improve line conditions.)
**Read the Loop Current & Circuit Loss Technical Bulletin at  [http://www.sandman.com/loopcur.html#CircuitLoss http://www.sandman.com/loopcur.html#CircuitLoss]
**Line conditioning apparatus is available at Sandman.com. Use the link above to reach this useful site.
*Disadvantages of analog lines are:
**Low technical capabilities
**Mediocre 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

== Extensions – How many extensions and what models? ==

*Physically review each location where a telephone will be installed
*Become aware of the person that will use the telephone and grasp challenges that this person will encounter – will this user require additional training
*Prepare to train users – on-line training is available at IPitomy Documentation Repository: [https://skydrive.live.com/?cid=a31a942025b325ce&id=A31A942025B325CE%21104 https://skydrive.live.com/?cid=a31a942025b325ce&id=A31A942025B325CE%21104].

= Network Readiness =

The network is the ''Backplane'' of the IP PBX. In old telephony solutions (TDM, Switched Matrix) the backplane was designed and controlled by the PBX manufacturer – not so in IP Telephony. The network MUST be in pristine condition to allow use with voice. Voice traffic MUST 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 voice quality from these issues.

== Cabling – 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.

== WAN – ISP ==

*Who is the Internet Service Provider at the site?
*Should this change?
*Are there cost advantages that can be offered in association to ISP and Trunks (SIP or PRI)?
*What is the WAN connection bandwidth performance?<ul style="list-style-type:circle;">
<li>A good simple tester: [http://speedtest.net/ http://speedtest.net/]</li>
<li>Keep in mind that ISP bandwidth testers are momentary snap-shots of the performance of your WAN (Internet) connection.</li>
<li>Several tests should be performed during the hours of operation at the site where the IP PBX will be installed.</li>
<li>Network performance software is available for network monitoring over extended periods. (Google it)</li>
</ul></ul>

== Router ==

*What kind of Router is in use?
*Are there multiple Public IP Addresses<ul style="list-style-type:circle;">
<li>What is/are the Public IP Address(es)</li>
</ul><li>Is this router on the list of Reportedly Works Well routers on the IPitomy web site? [http://www.ipitomy.com/webrelease/IPitomy/Router%20Compatibility%20Guide.pdf http://www.ipitomy.com/webrelease/IPitomy/Router%20Compatibility%20Guide.pdf]</li>
<li>Can you use your own router in Gateway mode to allow all voice traffic to be routed through that router versus routing all data and voice traffic through the router on site?</li>
<li>Who is in charge of the router? (Person, IT Dept?)<ul style="list-style-type:circle;">
<li>If IT Dept. controls the router and/or Switches on the site you MUST become best of buddies with the group of people. They WILL impact the performance of the voice traffic on the network.</li>
</ul></li>
<li>What device will serve DHCP?</li>
<li>Can you take over control of the router<ul style="list-style-type:circle;">
<li>If so, replace it with one of those on the IPitomy list that you have tested and found to perform as you intend.</li>
</ul></li>
<li>Is the router capable of Port Forwarding?<ul style="list-style-type:circle;">
<li>The router will need this function. The more robust the better. It is desirable to set Port Forward rules to receive packets on any port designation given by the sending device and then to deliver these packets to any LAN IP Address with a unique Port assignment.</li>
</ul></li>
<li>What is the access (User Name & Password) of the router?</li>
<li>How many concurrent voice traffic sessions is this router capable of NAT’ing? (NAT is Network Address Translation.) This is essential to the ability of a router to handle Remote Telephones.</li>
<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 information.</li>
</ul></li>
<li>Is an ALG (Application Layer Gateway) present on the router?<ul style="list-style-type:circle;">
<li>Is the ALG enabled?</li>
<li>If so, this will need to be disabled or;</li>
<li>…complex router programming is likely required to assure that no voice traffic is impeded by the operation of the ALG
*ALG’s often contribute to one-way audio issues.
*ALG’s often inhibit Remote Phone operation</li>
</li>

== Sonicwall® ==

Is a Sonicwall® present? If so, significant programming will be required to assure that no voice traffic (BOTH LAN and WAN traffic) is impeded.

*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>
</ul></ul>

== 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:

*Are they managed (programmable)?
*Are there enough ports to accommodate the number of telephones that will be installed?
*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></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

= Worksheet =

Complete the IPitomy Setup Worksheet located at: [http://www.ipitomy.com/webrelease/IPitomy/IP1100+/ http://www.ipitomy.com/webrelease/IPitomy/IP1100+/] each time that an IP PBX is to be installed. This form will help to organize the deployment of the PBX by aiding you to gather the information needed to program and prepare the site application. The form also provides a hard-copy backup of the database configuration.

= 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: [http://www.ipitomy.com/webrelease/SIP%20Providers/ http://www.ipitomy.com/webrelease/SIP%20Providers/]. 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:

*Never assume that a network is performing at its highest efficiency
*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 usually are internet-based and are therefore a non-guaranteed voice connection service due to fluctuations in bandwidth and WAN network performance. (Where PRI is a guaranteed service from the provider.)</li>
</ul></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.

Florist Application

2015-04-15T16:07:15Z

Nick Branica:

= Setting Up Retail Florist Overview =

The retail florist needs ring groups and menus set up so calls can be identified for the following reasons • For skills based routing. Staff has different levels of experience and limited to certain actions. For example, someone could be in the sales and customer service group and others may not be trained in customer service so they would only take sales calls. • Now owners on the other hand may use the find me follow me feature or they may be in a group so someone else could possibly answer their phone from another desk. This also is the same scenario with weddings, possibly a buying department and/or the design center. There are lots of choices here but I will break down the most common uses below. Since the IPitomy phone system works hand in hand with floral prodigy reporting, we need to add DID’s to the system. Our preferred phone service is Windstream. Therefore, Windstream will bring in an average of 60 DID’s. They then need to be pointed to the following ring groups that need to be created during installation or at any time in the future. The ones with * before them are optional depending on the operation

=== MAIN ===

Everyone must have the MAIN group. In our case it would be named MAIN. 25% of the numbers should point here including the main number phone number of the shop. This group will handle all calls coming in from print ads as well and would have a general message on hold uploaded that reflects the main points about the shop. Now before a call reaches this ring group you have options to set up menus to direct customers to a more specific call group. For example our menus are set to the following:

*Press 1 for Holiday order. When a customer comes in on the MAIN line and selects #1 for we know they are a high priority and ready to place an order. This group is only activated during holidays and announced on the voice greeting to press 1 if you want to place an order for the upcoming holiday which is also announced. Every sales agent in the sales groups are also in the HOLIDAY group. This also can have a special greeting when they are on hold that reflects the holiday. There would not be numbers pointing to this menu when there is no holiday.
*Press 2 for a regular order
*Press 3 for an existing order
*Press 4 for Weddings and Special Events  
*Press 5 for Accounting

=== WEBSITE ===

This group would need to have 50% of the numbers point to here. This is where we would pull phone numbers for GOOGLE, MSN, YELLOWPAGES.com or any online leads. When the customer clicks on the ad, this phone number will appear on the website and you know every caller in this group is looking at the website. We do not give them menu options in this group as we want them to stay here so we have trained agents to handle web related sales calls.

=== CUSTOMER SERVICE ===

This is usually just 1 phone number that people call with questions about their order and is also included in the menu of the MAIN group. It needs to be separated so you can have people trained accordingly to handle this group.

===  WEDDINGS ===

It is advised that 2-3 numbers are assigned here so leads can be tracked from the knot or wedding wire. This being in its own group allows for certain people to be added that could be designers or weddings and/or event staff. This group is also part of the MAIN menu in case a customer calls the main phone line.

== Additional groups ==

*VIP can be a special number that rings certain extension and even cell phones. This could be phone numbers given to all funeral homes, commercial standing order accounts or any special VIP that you want them to bypass the prompts and get your best people. This can also forward to a different destination. So for example, if you gave this number to brides and if they called after hours and this line forwarded to someone dedicated to be on call, you would be able to give them priority service. It is a selling point at Beneva that our brides have access to our staff anytime then need us. 
*Yellow Pages or another location you may have or a business you may have purchased. You can build groups depending on your needs to accommodate any experience you want to create. This way you control the Message on hold, the clerks available to take those calls and the experience the customer gets when they call.
*The options are almost endless as to what you can do. This is a starting point. The groups numbered 1-4 should be set up by default. Considering 1 number for Customer Service and a few in weddings and events so basically this leaves you with about 20% extra numbers. I woud say if they don’t have a place to point them just send the overflow to the website group as this is going to be the area to most likely grow.

The MAIN number will have the additional numbers used for ad campaigns in print, newspaper brochures, catalogs etc…

Remember, every call in campaign will also read the name of the campaign on the indicator on the Q Manager screen. The groups are for skilled based routing and creating the experience you wish for your customer to receive when they call your shop.

= Integration Overview =

There are Four major components to the IPitomy Floral application.  

#The '''IPitomy PBX''' which provides complete PBX services for the customers business, as well as Voice mail, Automated Attendant, ACD and all other PBX Services.
#* The PBX is installed along with the appropriate number of telephones.
#*The Florist application will require setup of Automated Attendants based on input from the customer
#*Groups will need to be setup with users/agents based on the customers requirements.
#*The IP address of the RTI Server is required to create a URL that will send the caller ID to the RTI Point of Sales terminal once the call is answered.  This URL will need to be programmedinto the PBX and the PBX will then distribute the URL to the Q Mangers when they are being setup.  The Caller ID is sent to the RTI treminal, along with other call information and is used by the user to search for the customers account.  This works similar to a screen pop, but the RTI server requires that the uaer/agent press a key in order to initiate the search.
#'''IPitomy Q Manager''' is desktop management software that resides on each users desktop.  This allows the users to monitor the call activity, see the status of other users, text message between users and much more. Q Manager is required to send the user/agents inbound Call ID to the RTI/Teleflora Point of Sale terminal.  make sure that each desktop has the URL loaded from the PBX in the CTI setup screen.
#'''RTI Server'''.  This is completely maintained by RTI.  The integration with the RTI server is already packaged up in the IPitomy Server.  What is needed for the RTI server is the LAN IP Address so it can be programmed into the IPitomy server so the PBX knows where the RTI server is located on the network.  The RTI server should be updated to be capable of integrating with IPitomy.  If it is not, you can proceed with the install as though it is ready.  Once the servier has been updated and is integrated with IPitomy, then everything should function normally provided there have been no major changes to the network topology.
#'''The Floral prodigy Web Site'''.  This collects all of the call data from the PBX and combines it with the order data from RTI.  This is a remote hosted server that is not on site.  All that needs to be done here is to make sure that there is a port on the router that is forwarded to port 80 on the PBX.  The Floral Prodigy Server automatically will retieve the call records from the IPitomy  PBX.  Care musrt be take to not add the IP Address of the Floral Prodigy server to the access control list so it is not banned from communicating with the IPitomy PBX.

Follow the instructions carefully and test each component to make sure it is functioning properly the way the customer needs it.

= Floral Prodigy Reports Package =

In order to set up Floral Prodigy two things need to be enabled. The IPitomy program must be setup to communicate with the Teleflora RTI program. IPitomy Tech Support can help with this. Be prepared to have the IP address of the PBX, Port 22 and port 80 forwarded to the PBX. You will need to provide the external IP address to the Floral Prodigy setup administrator and the IP address of the RTI system to the IPitomy technical support engineer to set up RTI. RTI will also need to enable your system for integration with the IPitomy and Floral Prodigy applications.

From the information we have gotten from the techs at Floral Prodigy, the following string is what works to poll our PBX for SMDR information:

 <nowiki>http://<PBX IP ADDRESS>/ippbx/exportCSV.php?stime=&etime=&type=CSV</nowiki>

You will need to set the stime value as the epoc time you want the report to start. This will ensure you are not pulling reports that repeat data from previous days. You do not need to set etime, but you can if you want to stop at a certain point, otherwise it will pull all records beginning at the stime value to current. NOTE: If there is any confusion, have Floral Prodigy check how they have other florist sites configured to work with IPitomy, as the process is the same.

It is not required, but if you enable SMDR Access Control List (navigate to PBX Setup=>Services and click the button for SMDR Access) you will need toadd the following IP ranges given to us by Gravity Free:

*64.156.29.0/24
*207.36.190.0/24

Do not enable SMDR Authentication. This should be set to No. This will allow the PBX to communicate to the Floral Prodigy servers in regards to SMDR.

{{:RTI Integration}}

Florist Application

2015-04-15T15:39:00Z

Nick Branica:

= Setting Up Retail Florist Overview =

The retail florist needs ring groups and menus set up so calls can be identified for the following reasons • For skills based routing. Staff has different levels of experience and limited to certain actions. For example, someone could be in the sales and customer service group and others may not be trained in customer service so they would only take sales calls. • Now owners on the other hand may use the find me follow me feature or they may be in a group so someone else could possibly answer their phone from another desk. This also is the same scenario with weddings, possibly a buying department and/or the design center. There are lots of choices here but I will break down the most common uses below. Since the IPitomy phone system works hand in hand with floral prodigy reporting, we need to add DID’s to the system. Our preferred phone service is Windstream. Therefore, Windstream will bring in an average of 60 DID’s. They then need to be pointed to the following ring groups that need to be created during installation or at any time in the future. The ones with * before them are optional depending on the operation

=== MAIN ===

Everyone must have the MAIN group. In our case it would be named MAIN. 25% of the numbers should point here including the main number phone number of the shop. This group will handle all calls coming in from print ads as well and would have a general message on hold uploaded that reflects the main points about the shop. Now before a call reaches this ring group you have options to set up menus to direct customers to a more specific call group. For example our menus are set to the following:

*Press 1 for Holiday order. When a customer comes in on the MAIN line and selects #1 for we know they are a high priority and ready to place an order. This group is only activated during holidays and announced on the voice greeting to press 1 if you want to place an order for the upcoming holiday which is also announced. Every sales agent in the sales groups are also in the HOLIDAY group. This also can have a special greeting when they are on hold that reflects the holiday. There would not be numbers pointing to this menu when there is no holiday.
*Press 2 for a regular order
*Press 3 for an existing order
*Press 4 for Weddings and Special Events  
*Press 5 for Accounting

=== WEBSITE ===

This group would need to have 50% of the numbers point to here. This is where we would pull phone numbers for GOOGLE, MSN, YELLOWPAGES.com or any online leads. When the customer clicks on the ad, this phone number will appear on the website and you know every caller in this group is looking at the website. We do not give them menu options in this group as we want them to stay here so we have trained agents to handle web related sales calls.

=== CUSTOMER SERVICE ===

This is usually just 1 phone number that people call with questions about their order and is also included in the menu of the MAIN group. It needs to be separated so you can have people trained accordingly to handle this group.

===  WEDDINGS ===

It is advised that 2-3 numbers are assigned here so leads can be tracked from the knot or wedding wire. This being in its own group allows for certain people to be added that could be designers or weddings and/or event staff. This group is also part of the MAIN menu in case a customer calls the main phone line.

== Additional groups ==

*VIP can be a special number that rings certain extension and even cell phones. This could be phone numbers given to all funeral homes, commercial standing order accounts or any special VIP that you want them to bypass the prompts and get your best people. This can also forward to a different destination. So for example, if you gave this number to brides and if they called after hours and this line forwarded to someone dedicated to be on call, you would be able to give them priority service. It is a selling point at Beneva that our brides have access to our staff anytime then need us. 
*Yellow Pages or another location you may have or a business you may have purchased. You can build groups depending on your needs to accommodate any experience you want to create. This way you control the Message on hold, the clerks available to take those calls and the experience the customer gets when they call.
*The options are almost endless as to what you can do. This is a starting point. The groups numbered 1-4 should be set up by default. Considering 1 number for Customer Service and a few in weddings and events so basically this leaves you with about 20% extra numbers. I woud say if they don’t have a place to point them just send the overflow to the website group as this is going to be the area to most likely grow.

The MAIN number will have the additional numbers used for ad campaigns in print, newspaper brochures, catalogs etc…

Remember, every call in campaign will also read the name of the campaign on the indicator on the Q Manager screen. The groups are for skilled based routing and creating the experience you wish for your customer to receive when they call your shop.

= Integration Overview =

There are Four major components to the IPitomy Floral application.  

#The '''IPitomy PBX''' which provides complete PBX services for the customers business, as well as Voice mail, Automated Attendant, ACD and all other PBX Services.
#* The PBX is installed along with the appropriate number of telephones.
#*The Florist application will require setup of Automated Attendants based on input from the customer
#*Groups will need to be setup with users/agents based on the customers requirements.
#*The IP address of the RTI Server is required to create a URL that will send the caller ID to the RTI Point of Sales terminal once the call is answered.  This URL will need to be programmedinto the PBX and the PBX will then distribute the URL to the Q Mangers when they are being setup.
#'''IPitomy Q Manager''' is desktop management software that resides on each users desktop.  This allows the users to monitor the call activity, see the status of other users, text message between users and much more. Q Manager is required to send the user/agents inbound Call ID to the RTI/Teleflora Point of Sale terminal.  make sure that each desktop has the URL loaded from the PBX in the CTI setup screen.
#'''RTI Server'''.  This is completely maintained by RTI.  The integration with the RTI server is already packaged up in the IPitomy Server.  What is needed for the RTI server is the LAN IP Address so it can be programmed into the IPitomy server so the PBX knows where the RTI server is located on the network.  The RTI server should be updated to be capable of integrating with IPitomy.  If it is not, you can proceed with the install as though it is ready.  Once the servier has been updated and is integrated with IPitomy, then everything should function normally provided there have been no major changes to the network topology.
#'''The Floral prodigy Web Site'''.  This collects all of the call data from the PBX and combines it with the order data from RTI.  This is a remote hosted server that is not on site.  All that needs to be done here is to make sure that there is a port on the router that is forwarded to port 80 on the PBX.  The Floral Prodigy Server automatically will retieve the call records from the IPitomy  PBX.  Care musrt be take to not add the IP Address of the Floral Prodigy server to the access control list so it is not banned from communicating with the IPitomy PBX.

Follow the instructions carefully and test each component to make sure it is functioning properly the way the customer needs it.

= Floral Prodigy Reports Package =

In order to set up Floral Prodigy two things need to be enabled. The IPitomy program must be setup to communicate with the Teleflora RTI program. IPitomy Tech Support can help with this. Be prepared to have the IP address of the PBX, Port 22 and port 80 forwarded to the PBX. You will need to provide the external IP address to the Floral Prodigy setup administrator and the IP address of the RTI system to the IPitomy technical support engineer to set up RTI. RTI will also need to enable your system for integration with the IPitomy and Floral Prodigy applications.

From the information we have gotten from the techs at Floral Prodigy, the following string is what works to poll our PBX for SMDR information:

 <nowiki>http://<PBX IP ADDRESS>/ippbx/exportCSV.php?stime=&etime=&type=CSV</nowiki>

You will need to set the stime value as the epoc time you want the report to start. This will ensure you are not pulling reports that repeat data from previous days. You do not need to set etime, but you can if you want to stop at a certain point, otherwise it will pull all records beginning at the stime value to current. NOTE: If there is any confusion, have Floral Prodigy check how they have other florist sites configured to work with IPitomy, as the process is the same.

It is not required, but if you enable SMDR Access Control List (navigate to PBX Setup=>Services and click the button for SMDR Access) you will need toadd the following IP ranges given to us by Gravity Free:

*64.156.29.0/24
*207.36.190.0/24

Do not enable SMDR Authentication. This should be set to No. This will allow the PBX to communicate to the Floral Prodigy servers in regards to SMDR.

{{:RTI Integration}}

Routing Ops

2015-04-15T14:30:53Z

Nick Branica:

Advanced Call Routing - Routing Ops is a licensed feature that allows inbound calls to route based on any one of the following

*Inbound CID
*Number Dialed
*Collected Digits

'''Advanced Call Routing''' - Routing Ops can be used in conjunction with a Group as a filter app, or you can set a Routing Op entry as a destination anywhere in the system. A Routing app basically consists of one or more ‘expressions’ that resolve to be either True or False. The Operation evaluates each expression sequentially in the defined order (priority). If an expression resolves as True, the caller is immediately routed to the configured route destination for that expression. If the condition is False, the Operation evaluates the next expression. If all expression evaluate to False, the Operation will perform the default behavior based on how the App was invoked (filter or destination). Note: A disabled Routing Operation skips all expressions and performs the ‘default behavior’ based on how the App was invoked.

The part Number to Purchase this License - advanced_cr

 [[File:Application-Routing Ops-.jpg|File:Application-Routing Ops-.jpg]]

== Group Method ==

If the app is invoked as a Filter for a Group, the default behavior is to do nothing (i.e. stay in the group as if the filter was never invoked). Specifically, the ‘Default Destination’ field configured on the Route Operations Edit page is ignored.

== Destination Method ==

If the app is invoked as a ‘Destination’, the default behavior is to route to the 'Default Destination’ field configured on the Route Operations Edit page.

== Building a Routing Op ==

[[File:Application-Routing Ops-Edit.jpg|File:Application-Routing Ops-Edit.jpg]]

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Name 
| Name of the Routing Op.  You should rename to something clear that explains what the route does. 
| Routing Application 
|-
| Operation Type 
| Can be set to Match List of Numbers (single variable, exact match) or Table Lookup (multivariable, exact match or mathmatical functions).  Once set, this cannot be changed. 
| Match List of Numbers. 
|-
| Enabled 
| Toggle the Op ON or OFF 
| No 
|-
| Prompt & Collect Digits 
| Defines if the system will prompt the caller to enter digits before evaluating expressions.  When you select Yes, additional fields are displayed. 
| No 
|-
| Default Destination 
| Where a call will go if it does not meet a filtered status. (Destination method only, not needed for Group method) 
| None 
|-
| Prompt to Play 
| Set this to a recorded prompt instructing the caller what digits to enter.  Only used when Prompt & Collect Digits is set to Yes. 
| 1-sec-silence 
|-
| Number of Digits to Collect 
|
Total number of digits to collect. Only used when Prompt & Collect Digits is set to Yes.

| 4 
|-
|
Number of seconds to wait if no digits are entered

| How long the system will wait for digits to be entered. Only used when Prompt & Collect Digits is set to Yes. 
| 10 
|-
|
Number of Attempts to Allow Before Routing to the Failover Destination

| How many tries the system will give the caller to enter their digits before routing to the Failover Destination. Only used when Prompt & Collect Digits is set to Yes. 
| 3 
|-
| Failover Destination 
| Where the call routes if the user does not enter digits 
| None 
|-
| Available Group Selections 
| Groups that can be chosen to use the Group method of Routing Ops 
| All Unselected Groups 
|-
| Selected Groups 
| List of Groups that have been added to this Routing Op 
| None 
|}

== Configuring Expressions: Match List Method ==

[[File:Application-Routing Ops-Match List-Add Expression.jpg|File:Application-Routing Ops-Match List-Add Expression.jpg]]

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Input Type 
| Caller ID: Route by the CID of the caller Dialed Number: Route by the DID dialed by the caller Collected Digits: Route by digits entered by the caller after connecting to the PBX 
| Caller ID 
|-
| Use the Right Most 
| Total number of digits to use in the input parameter 
| 10 
|-
| Starting in position X 
| Starting from the left, how many digits to skip before reading to compare to parameter 
| 0 
|-
| Collect X Digits 
| How many digits to read moving right from the starting positition 
| 10 
|-
| Match Input Parameter 
| This is where you will enter the values that will be compared, resulting in a True or False 
| None 
|-
| If the input parameter X matches a member of the list 
| This can be set to Matches or Does Not Match, depending on the functionality you want to result in a re-routing of the call. 
| Matches 
|-
| Route to Destination 
| Set this to any destination on the PBX, and will trigger when the input data matches the parameters configured. 
| None 
|}

== Configuring Expressions: Table Lookup Method ==

[[File:Application-Routing Ops-Edit-TableLookup.jpg|File:Application-Routing Ops-Edit-TableLookup.jpg]]

*Once the Routing Op has been saved as Table Lookup method, a button reading Manage Data will appear.  Click this to access the advanced methods of defining expressions available to the Table Lookup operation.

[[File:Application-Routing Ops-Table-Creation.jpg|File:Application-Routing Ops-Table-Creation.jpg]]

*Clicking Manage Data will bring up options to manually enter data into a new table, or to load a CSV file with the parameters you wish to use.

[[File:Application-Routing Ops-Table-Creation-Manual.jpg|File:Application-Routing Ops-Table-Creation-Manual.jpg]]

*Selecting Manual method will bring you to the next page, allowing you to define if the table created should have 1, 2, or 3 columns.

[[File:Application-Routing Ops-Table-Creation-Manual-Populate.jpg|File:Application-Routing Ops-Table-Creation-Manual-Populate.jpg]]

*Which takes you to the next page, allowing you to define the expressions in each column, as well as the ability to create or remove rows, allowing you to add or delete parameters.

[[File:Application-Routing Ops-Table-Creation-CSV.jpg|File:Application-Routing Ops-Table-Creation-CSV.jpg]]

*If you choose to load a CSV file, the page above will display.  With this you get to define how many columns the table will have, where to import each CSV column into the Routing Ops table, and then allowing you to browse to and load the file.

[[File:Application-Routing Ops-Table-Add Expression.jpg|File:Application-Routing Ops-Table-Add Expression.jpg]]

*Regardless of whether the manual or CSV method were used to create the table, the same page will display when you back out to the Edit page for the Routing Op and click Expressions.  The unique fields to this page are defined below.

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Get value in Column X 
| Define what colume to refer to for this comparison.  Values can be 1, 2, or 3 
| 1 
|-
| as an X value 
| Are we comparing a Text value or a Number 
| Text 
|-
| Use the Right Most X digits 
| Total number of digits to use in the input parameter 
| 10 
|-
| Starting in position X 
| Starting from the left, how many digits to skip before reading to compare to parameter 
| 0 
|-
| Collect X Digits 
| How many digits to read moving right from the starting positition 
| 10 
|-
| Get value in column X as an Y value for parameter 2 
| Define what column to read, and what type of data it will be reading for comparison of parameter 2.  Column can be 1, 2, or 3. Type can be Number or Text. 
| 2, Text 
|-
| Get value in column X as an Y value for parameter 3 
| Define what column to read, and what type of data it will be reading for comparison of parameter 3.  Column can be 1, 2, or 3. Type can be Number or Text. 
| 3, Text 
|-
| If the Caller Input [Dropdown] a value in the Lookup Column And 
| Is equal to OR Is not equal to are the options here, defining if you take action when there is or is not a match. 
| Is Equal to 
|-
| Table parameter two [Dropdown] ________ 
| Is equal to OR Is not equal to are the options here, defining if you take action when there is or is not a match.  The text field is where you enter the expression to compare to. 
| Is Equal to 
|-
| AND/OR 
| Choose either And OR Or.  And means both must match, or means either can match. 
| AND 
|-
| Table parameter three [DropDown] ________ 
| Is equal to OR Is not equal to are the options for the dropdown, defining if you take action when there is or is not a match.  The text field is where you enter the expression to compare to. 
| 
|-
| Route To Destination 
| Where to send the call if all aprameter comparisons match 
| None 
|}

Routing Ops

2015-04-15T14:29:54Z

Nick Branica:

Advanced Call Routing - Routing Ops is a licensed feature that allows inbound calls to route based on any one of the following

*Inbound CID
*Number Dialed
*Collected Digits

'''Advanced Call Routing''' - Routing Ops can be used in conjunction with a Group as a filter app, or you can set a Routing Op entry as a destination anywhere in the system. A Routing app basically consists of one or more ‘expressions’ that resolve to be either True or False. The Operation evaluates each expression sequentially in the defined order (priority). If an expression resolves as True, the caller is immediately routed to the configured route destination for that expression. If the condition is False, the Operation evaluates the next expression. If all expression evaluate to False, the Operation will perform the default behavior based on how the App was invoked (filter or destination). Note: A disabled Routing Operation skips all expressions and performs the ‘default behavior’ based on how the App was invoked.

part Number to Purchase this License - advanced_cr

 [[File:Application-Routing Ops-.jpg|File:Application-Routing Ops-.jpg]]

== Group Method ==

If the app is invoked as a Filter for a Group, the default behavior is to do nothing (i.e. stay in the group as if the filter was never invoked). Specifically, the ‘Default Destination’ field configured on the Route Operations Edit page is ignored.

== Destination Method ==

If the app is invoked as a ‘Destination’, the default behavior is to route to the 'Default Destination’ field configured on the Route Operations Edit page.

== Building a Routing Op ==

[[File:Application-Routing Ops-Edit.jpg|File:Application-Routing Ops-Edit.jpg]]

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Name 
| Name of the Routing Op.  You should rename to something clear that explains what the route does. 
| Routing Application 
|-
| Operation Type 
| Can be set to Match List of Numbers (single variable, exact match) or Table Lookup (multivariable, exact match or mathmatical functions).  Once set, this cannot be changed. 
| Match List of Numbers. 
|-
| Enabled 
| Toggle the Op ON or OFF 
| No 
|-
| Prompt & Collect Digits 
| Defines if the system will prompt the caller to enter digits before evaluating expressions.  When you select Yes, additional fields are displayed. 
| No 
|-
| Default Destination 
| Where a call will go if it does not meet a filtered status. (Destination method only, not needed for Group method) 
| None 
|-
| Prompt to Play 
| Set this to a recorded prompt instructing the caller what digits to enter.  Only used when Prompt & Collect Digits is set to Yes. 
| 1-sec-silence 
|-
| Number of Digits to Collect 
|
Total number of digits to collect. Only used when Prompt & Collect Digits is set to Yes.

| 4 
|-
|
Number of seconds to wait if no digits are entered

| How long the system will wait for digits to be entered. Only used when Prompt & Collect Digits is set to Yes. 
| 10 
|-
|
Number of Attempts to Allow Before Routing to the Failover Destination

| How many tries the system will give the caller to enter their digits before routing to the Failover Destination. Only used when Prompt & Collect Digits is set to Yes. 
| 3 
|-
| Failover Destination 
| Where the call routes if the user does not enter digits 
| None 
|-
| Available Group Selections 
| Groups that can be chosen to use the Group method of Routing Ops 
| All Unselected Groups 
|-
| Selected Groups 
| List of Groups that have been added to this Routing Op 
| None 
|}

== Configuring Expressions: Match List Method ==

[[File:Application-Routing Ops-Match List-Add Expression.jpg|File:Application-Routing Ops-Match List-Add Expression.jpg]]

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Input Type 
| Caller ID: Route by the CID of the caller Dialed Number: Route by the DID dialed by the caller Collected Digits: Route by digits entered by the caller after connecting to the PBX 
| Caller ID 
|-
| Use the Right Most 
| Total number of digits to use in the input parameter 
| 10 
|-
| Starting in position X 
| Starting from the left, how many digits to skip before reading to compare to parameter 
| 0 
|-
| Collect X Digits 
| How many digits to read moving right from the starting positition 
| 10 
|-
| Match Input Parameter 
| This is where you will enter the values that will be compared, resulting in a True or False 
| None 
|-
| If the input parameter X matches a member of the list 
| This can be set to Matches or Does Not Match, depending on the functionality you want to result in a re-routing of the call. 
| Matches 
|-
| Route to Destination 
| Set this to any destination on the PBX, and will trigger when the input data matches the parameters configured. 
| None 
|}

== Configuring Expressions: Table Lookup Method ==

[[File:Application-Routing Ops-Edit-TableLookup.jpg|File:Application-Routing Ops-Edit-TableLookup.jpg]]

*Once the Routing Op has been saved as Table Lookup method, a button reading Manage Data will appear.  Click this to access the advanced methods of defining expressions available to the Table Lookup operation.

[[File:Application-Routing Ops-Table-Creation.jpg|File:Application-Routing Ops-Table-Creation.jpg]]

*Clicking Manage Data will bring up options to manually enter data into a new table, or to load a CSV file with the parameters you wish to use.

[[File:Application-Routing Ops-Table-Creation-Manual.jpg|File:Application-Routing Ops-Table-Creation-Manual.jpg]]

*Selecting Manual method will bring you to the next page, allowing you to define if the table created should have 1, 2, or 3 columns.

[[File:Application-Routing Ops-Table-Creation-Manual-Populate.jpg|File:Application-Routing Ops-Table-Creation-Manual-Populate.jpg]]

*Which takes you to the next page, allowing you to define the expressions in each column, as well as the ability to create or remove rows, allowing you to add or delete parameters.

[[File:Application-Routing Ops-Table-Creation-CSV.jpg|File:Application-Routing Ops-Table-Creation-CSV.jpg]]

*If you choose to load a CSV file, the page above will display.  With this you get to define how many columns the table will have, where to import each CSV column into the Routing Ops table, and then allowing you to browse to and load the file.

[[File:Application-Routing Ops-Table-Add Expression.jpg|File:Application-Routing Ops-Table-Add Expression.jpg]]

*Regardless of whether the manual or CSV method were used to create the table, the same page will display when you back out to the Edit page for the Routing Op and click Expressions.  The unique fields to this page are defined below.

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Get value in Column X 
| Define what colume to refer to for this comparison.  Values can be 1, 2, or 3 
| 1 
|-
| as an X value 
| Are we comparing a Text value or a Number 
| Text 
|-
| Use the Right Most X digits 
| Total number of digits to use in the input parameter 
| 10 
|-
| Starting in position X 
| Starting from the left, how many digits to skip before reading to compare to parameter 
| 0 
|-
| Collect X Digits 
| How many digits to read moving right from the starting positition 
| 10 
|-
| Get value in column X as an Y value for parameter 2 
| Define what column to read, and what type of data it will be reading for comparison of parameter 2.  Column can be 1, 2, or 3. Type can be Number or Text. 
| 2, Text 
|-
| Get value in column X as an Y value for parameter 3 
| Define what column to read, and what type of data it will be reading for comparison of parameter 3.  Column can be 1, 2, or 3. Type can be Number or Text. 
| 3, Text 
|-
| If the Caller Input [Dropdown] a value in the Lookup Column And 
| Is equal to OR Is not equal to are the options here, defining if you take action when there is or is not a match. 
| Is Equal to 
|-
| Table parameter two [Dropdown] ________ 
| Is equal to OR Is not equal to are the options here, defining if you take action when there is or is not a match.  The text field is where you enter the expression to compare to. 
| Is Equal to 
|-
| AND/OR 
| Choose either And OR Or.  And means both must match, or means either can match. 
| AND 
|-
| Table parameter three [DropDown] ________ 
| Is equal to OR Is not equal to are the options for the dropdown, defining if you take action when there is or is not a match.  The text field is where you enter the expression to compare to. 
| 
|-
| Route To Destination 
| Where to send the call if all aprameter comparisons match 
| None 
|}

Routing Ops

2015-04-15T14:27:59Z

Nick Branica:

Advanced Call Routing - Routing Ops is a licensed feature that allows inbound calls to route based on any one of the following

*Inbound CID
*Number Dialed
*Collected Digits

Routing Ops can be used in conjunction with a Group as a filter app, or you can set a Routing Op entry as a destination anywhere in the system. A Routing app basically consists of one or more ‘expressions’ that resolve to be either True or False. The Operation evaluates each expression sequentially in the defined order (priority). If an expression resolves as True, the caller is immediately routed to the configured route destination for that expression. If the condition is False, the Operation evaluates the next expression. If all expression evaluate to False, the Operation will perform the default behavior based on how the App was invoked (filter or destination). Note: A disabled Routing Operation skips all expressions and performs the ‘default behavior’ based on how the App was invoked.

part Number to Purchase this License - advanced_cr

 [[File:Application-Routing Ops-.jpg|File:Application-Routing Ops-.jpg]]

== Group Method ==

If the app is invoked as a Filter for a Group, the default behavior is to do nothing (i.e. stay in the group as if the filter was never invoked). Specifically, the ‘Default Destination’ field configured on the Route Operations Edit page is ignored.

== Destination Method ==

If the app is invoked as a ‘Destination’, the default behavior is to route to the 'Default Destination’ field configured on the Route Operations Edit page.

== Building a Routing Op ==

[[File:Application-Routing Ops-Edit.jpg|File:Application-Routing Ops-Edit.jpg]]

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Name 
| Name of the Routing Op.  You should rename to something clear that explains what the route does. 
| Routing Application 
|-
| Operation Type 
| Can be set to Match List of Numbers (single variable, exact match) or Table Lookup (multivariable, exact match or mathmatical functions).  Once set, this cannot be changed. 
| Match List of Numbers. 
|-
| Enabled 
| Toggle the Op ON or OFF 
| No 
|-
| Prompt & Collect Digits 
| Defines if the system will prompt the caller to enter digits before evaluating expressions.  When you select Yes, additional fields are displayed. 
| No 
|-
| Default Destination 
| Where a call will go if it does not meet a filtered status. (Destination method only, not needed for Group method) 
| None 
|-
| Prompt to Play 
| Set this to a recorded prompt instructing the caller what digits to enter.  Only used when Prompt & Collect Digits is set to Yes. 
| 1-sec-silence 
|-
| Number of Digits to Collect 
|
Total number of digits to collect. Only used when Prompt & Collect Digits is set to Yes.

| 4 
|-
|
Number of seconds to wait if no digits are entered

| How long the system will wait for digits to be entered. Only used when Prompt & Collect Digits is set to Yes. 
| 10 
|-
|
Number of Attempts to Allow Before Routing to the Failover Destination

| How many tries the system will give the caller to enter their digits before routing to the Failover Destination. Only used when Prompt & Collect Digits is set to Yes. 
| 3 
|-
| Failover Destination 
| Where the call routes if the user does not enter digits 
| None 
|-
| Available Group Selections 
| Groups that can be chosen to use the Group method of Routing Ops 
| All Unselected Groups 
|-
| Selected Groups 
| List of Groups that have been added to this Routing Op 
| None 
|}

== Configuring Expressions: Match List Method ==

[[File:Application-Routing Ops-Match List-Add Expression.jpg|File:Application-Routing Ops-Match List-Add Expression.jpg]]

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Input Type 
| Caller ID: Route by the CID of the caller Dialed Number: Route by the DID dialed by the caller Collected Digits: Route by digits entered by the caller after connecting to the PBX 
| Caller ID 
|-
| Use the Right Most 
| Total number of digits to use in the input parameter 
| 10 
|-
| Starting in position X 
| Starting from the left, how many digits to skip before reading to compare to parameter 
| 0 
|-
| Collect X Digits 
| How many digits to read moving right from the starting positition 
| 10 
|-
| Match Input Parameter 
| This is where you will enter the values that will be compared, resulting in a True or False 
| None 
|-
| If the input parameter X matches a member of the list 
| This can be set to Matches or Does Not Match, depending on the functionality you want to result in a re-routing of the call. 
| Matches 
|-
| Route to Destination 
| Set this to any destination on the PBX, and will trigger when the input data matches the parameters configured. 
| None 
|}

== Configuring Expressions: Table Lookup Method ==

[[File:Application-Routing Ops-Edit-TableLookup.jpg|File:Application-Routing Ops-Edit-TableLookup.jpg]]

*Once the Routing Op has been saved as Table Lookup method, a button reading Manage Data will appear.  Click this to access the advanced methods of defining expressions available to the Table Lookup operation.

[[File:Application-Routing Ops-Table-Creation.jpg|File:Application-Routing Ops-Table-Creation.jpg]]

*Clicking Manage Data will bring up options to manually enter data into a new table, or to load a CSV file with the parameters you wish to use.

[[File:Application-Routing Ops-Table-Creation-Manual.jpg|File:Application-Routing Ops-Table-Creation-Manual.jpg]]

*Selecting Manual method will bring you to the next page, allowing you to define if the table created should have 1, 2, or 3 columns.

[[File:Application-Routing Ops-Table-Creation-Manual-Populate.jpg|File:Application-Routing Ops-Table-Creation-Manual-Populate.jpg]]

*Which takes you to the next page, allowing you to define the expressions in each column, as well as the ability to create or remove rows, allowing you to add or delete parameters.

[[File:Application-Routing Ops-Table-Creation-CSV.jpg|File:Application-Routing Ops-Table-Creation-CSV.jpg]]

*If you choose to load a CSV file, the page above will display.  With this you get to define how many columns the table will have, where to import each CSV column into the Routing Ops table, and then allowing you to browse to and load the file.

[[File:Application-Routing Ops-Table-Add Expression.jpg|File:Application-Routing Ops-Table-Add Expression.jpg]]

*Regardless of whether the manual or CSV method were used to create the table, the same page will display when you back out to the Edit page for the Routing Op and click Expressions.  The unique fields to this page are defined below.

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Get value in Column X 
| Define what colume to refer to for this comparison.  Values can be 1, 2, or 3 
| 1 
|-
| as an X value 
| Are we comparing a Text value or a Number 
| Text 
|-
| Use the Right Most X digits 
| Total number of digits to use in the input parameter 
| 10 
|-
| Starting in position X 
| Starting from the left, how many digits to skip before reading to compare to parameter 
| 0 
|-
| Collect X Digits 
| How many digits to read moving right from the starting positition 
| 10 
|-
| Get value in column X as an Y value for parameter 2 
| Define what column to read, and what type of data it will be reading for comparison of parameter 2.  Column can be 1, 2, or 3. Type can be Number or Text. 
| 2, Text 
|-
| Get value in column X as an Y value for parameter 3 
| Define what column to read, and what type of data it will be reading for comparison of parameter 3.  Column can be 1, 2, or 3. Type can be Number or Text. 
| 3, Text 
|-
| If the Caller Input [Dropdown] a value in the Lookup Column And 
| Is equal to OR Is not equal to are the options here, defining if you take action when there is or is not a match. 
| Is Equal to 
|-
| Table parameter two [Dropdown] ________ 
| Is equal to OR Is not equal to are the options here, defining if you take action when there is or is not a match.  The text field is where you enter the expression to compare to. 
| Is Equal to 
|-
| AND/OR 
| Choose either And OR Or.  And means both must match, or means either can match. 
| AND 
|-
| Table parameter three [DropDown] ________ 
| Is equal to OR Is not equal to are the options for the dropdown, defining if you take action when there is or is not a match.  The text field is where you enter the expression to compare to. 
| 
|-
| Route To Destination 
| Where to send the call if all aprameter comparisons match 
| None 
|}

Routing Ops

2015-04-15T14:26:10Z

Nick Branica:

Advanced Call Routing - Routing Ops is a licensed feature that allows inbound calls to route based on any one of the following

*Inbound CID
*Number Dialed
*Collected Digits

Routing Ops can be used in conjunction with a Group as a filter app, or you can set a Routing Op entry as a destination anywhere in the system. A Routing app basically consists of one or more ‘expressions’ that resolve to be either True or False. The Operation evaluates each expression sequentially in the defined order (priority). If an expression resolves as True, the caller is immediately routed to the configured route destination for that expression. If the condition is False, the Operation evaluates the next expression. If all expression evaluate to False, the Operation will perform the default behavior based on how the App was invoked (filter or destination). Note: A disabled Routing Operation skips all expressions and performs the ‘default behavior’ based on how the App was invoked.

 [[File:Application-Routing Ops-.jpg|File:Application-Routing Ops-.jpg]]

== Group Method ==

If the app is invoked as a Filter for a Group, the default behavior is to do nothing (i.e. stay in the group as if the filter was never invoked). Specifically, the ‘Default Destination’ field configured on the Route Operations Edit page is ignored.

== Destination Method ==

If the app is invoked as a ‘Destination’, the default behavior is to route to the 'Default Destination’ field configured on the Route Operations Edit page.

== Building a Routing Op ==

[[File:Application-Routing Ops-Edit.jpg|File:Application-Routing Ops-Edit.jpg]]

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Name 
| Name of the Routing Op.  You should rename to something clear that explains what the route does. 
| Routing Application 
|-
| Operation Type 
| Can be set to Match List of Numbers (single variable, exact match) or Table Lookup (multivariable, exact match or mathmatical functions).  Once set, this cannot be changed. 
| Match List of Numbers. 
|-
| Enabled 
| Toggle the Op ON or OFF 
| No 
|-
| Prompt & Collect Digits 
| Defines if the system will prompt the caller to enter digits before evaluating expressions.  When you select Yes, additional fields are displayed. 
| No 
|-
| Default Destination 
| Where a call will go if it does not meet a filtered status. (Destination method only, not needed for Group method) 
| None 
|-
| Prompt to Play 
| Set this to a recorded prompt instructing the caller what digits to enter.  Only used when Prompt & Collect Digits is set to Yes. 
| 1-sec-silence 
|-
| Number of Digits to Collect 
|
Total number of digits to collect. Only used when Prompt & Collect Digits is set to Yes.

| 4 
|-
|
Number of seconds to wait if no digits are entered

| How long the system will wait for digits to be entered. Only used when Prompt & Collect Digits is set to Yes. 
| 10 
|-
|
Number of Attempts to Allow Before Routing to the Failover Destination

| How many tries the system will give the caller to enter their digits before routing to the Failover Destination. Only used when Prompt & Collect Digits is set to Yes. 
| 3 
|-
| Failover Destination 
| Where the call routes if the user does not enter digits 
| None 
|-
| Available Group Selections 
| Groups that can be chosen to use the Group method of Routing Ops 
| All Unselected Groups 
|-
| Selected Groups 
| List of Groups that have been added to this Routing Op 
| None 
|}

== Configuring Expressions: Match List Method ==

[[File:Application-Routing Ops-Match List-Add Expression.jpg|File:Application-Routing Ops-Match List-Add Expression.jpg]]

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Input Type 
| Caller ID: Route by the CID of the caller Dialed Number: Route by the DID dialed by the caller Collected Digits: Route by digits entered by the caller after connecting to the PBX 
| Caller ID 
|-
| Use the Right Most 
| Total number of digits to use in the input parameter 
| 10 
|-
| Starting in position X 
| Starting from the left, how many digits to skip before reading to compare to parameter 
| 0 
|-
| Collect X Digits 
| How many digits to read moving right from the starting positition 
| 10 
|-
| Match Input Parameter 
| This is where you will enter the values that will be compared, resulting in a True or False 
| None 
|-
| If the input parameter X matches a member of the list 
| This can be set to Matches or Does Not Match, depending on the functionality you want to result in a re-routing of the call. 
| Matches 
|-
| Route to Destination 
| Set this to any destination on the PBX, and will trigger when the input data matches the parameters configured. 
| None 
|}

== Configuring Expressions: Table Lookup Method ==

[[File:Application-Routing Ops-Edit-TableLookup.jpg|File:Application-Routing Ops-Edit-TableLookup.jpg]]

*Once the Routing Op has been saved as Table Lookup method, a button reading Manage Data will appear.  Click this to access the advanced methods of defining expressions available to the Table Lookup operation.

[[File:Application-Routing Ops-Table-Creation.jpg|File:Application-Routing Ops-Table-Creation.jpg]]

*Clicking Manage Data will bring up options to manually enter data into a new table, or to load a CSV file with the parameters you wish to use.

[[File:Application-Routing Ops-Table-Creation-Manual.jpg|File:Application-Routing Ops-Table-Creation-Manual.jpg]]

*Selecting Manual method will bring you to the next page, allowing you to define if the table created should have 1, 2, or 3 columns.

[[File:Application-Routing Ops-Table-Creation-Manual-Populate.jpg|File:Application-Routing Ops-Table-Creation-Manual-Populate.jpg]]

*Which takes you to the next page, allowing you to define the expressions in each column, as well as the ability to create or remove rows, allowing you to add or delete parameters.

[[File:Application-Routing Ops-Table-Creation-CSV.jpg|File:Application-Routing Ops-Table-Creation-CSV.jpg]]

*If you choose to load a CSV file, the page above will display.  With this you get to define how many columns the table will have, where to import each CSV column into the Routing Ops table, and then allowing you to browse to and load the file.

[[File:Application-Routing Ops-Table-Add Expression.jpg|File:Application-Routing Ops-Table-Add Expression.jpg]]

*Regardless of whether the manual or CSV method were used to create the table, the same page will display when you back out to the Edit page for the Routing Op and click Expressions.  The unique fields to this page are defined below.

{| style="width: 500px" border="1" cellpadding="1" cellspacing="1"
|-
| '''Field''' 
| '''Function''' 
| '''Default''' 
|-
| Get value in Column X 
| Define what colume to refer to for this comparison.  Values can be 1, 2, or 3 
| 1 
|-
| as an X value 
| Are we comparing a Text value or a Number 
| Text 
|-
| Use the Right Most X digits 
| Total number of digits to use in the input parameter 
| 10 
|-
| Starting in position X 
| Starting from the left, how many digits to skip before reading to compare to parameter 
| 0 
|-
| Collect X Digits 
| How many digits to read moving right from the starting positition 
| 10 
|-
| Get value in column X as an Y value for parameter 2 
| Define what column to read, and what type of data it will be reading for comparison of parameter 2.  Column can be 1, 2, or 3. Type can be Number or Text. 
| 2, Text 
|-
| Get value in column X as an Y value for parameter 3 
| Define what column to read, and what type of data it will be reading for comparison of parameter 3.  Column can be 1, 2, or 3. Type can be Number or Text. 
| 3, Text 
|-
| If the Caller Input [Dropdown] a value in the Lookup Column And 
| Is equal to OR Is not equal to are the options here, defining if you take action when there is or is not a match. 
| Is Equal to 
|-
| Table parameter two [Dropdown] ________ 
| Is equal to OR Is not equal to are the options here, defining if you take action when there is or is not a match.  The text field is where you enter the expression to compare to. 
| Is Equal to 
|-
| AND/OR 
| Choose either And OR Or.  And means both must match, or means either can match. 
| AND 
|-
| Table parameter three [DropDown] ________ 
| Is equal to OR Is not equal to are the options for the dropdown, defining if you take action when there is or is not a match.  The text field is where you enter the expression to compare to. 
| 
|-
| Route To Destination 
| Where to send the call if all aprameter comparisons match 
| None 
|}

Directed Park

2015-03-13T20:53:52Z

Nick Branica:

The IPitomy directed park feature allows users to park a call and pick up the parked call on the same key. This is a standard feature in version 5.1.1. There are 20 park "extensions/orbits" 701 - 720. These park keys can be setup as a BLF key as usual. No changes to the park keys need to be made. You may choose to delete the "park" key to free up another button location on the users phone.

It is essential to set BLF transfer mode to "Attended" in the key programming "Display/Call Control" tab on Pone Settings.

The Directed Park feature can be disabled or enabled. If disabled, the standard IPitomy Dynamic parking will work as it always has.

To use the feature, a customer needs to be on a call. While on the call, press the park key/BLF you wish to park the caller on. The caller will be automatically transferred to the park extension/orbit. The BLF assigned to this extension/orbit will blink on all phones that have this programmed as a BLF. Simply press the blinking BLF and pick up the "Parked" call.

The directed park feature can fulfill requirements for users that need departmental parks so only assigned users will see that there is a caller parked in the park orbit. For instance, the receptionist can announce call parked for "parts" and any agent in the parts department will be able to pick up the parked call using the blinking BLF on their phone. Other users can pick up the parked call by dialing the pickup code ((() followed by the specific park orbit i.e. "99701" if 701 is the park orbit assigned to parts.

The feature works in V2 & V3 telephones. It is not possible for this feature to be used in V1 phones. For V1 phones, see below.

== Alternative parking programming for V1 phones ==

Sometimes an end user is not satisfied with the native Park feature found in the PBX. Instead of Parking calls to the first available orbit, they would rather have the ability to define where the call goes, parking and retrieving the call by pressing the same key and always knowing where its going. In an effort to accommodate this, we have come up with the following solution:

*Build a number of Extensions on the PBX that will be used as Directed Park, in this example we will use 801, 802, 803.
**Set the Busy, NoAnswer, and Unavailable Forwarding for these extensions to a destination that will be the failover if the call is not retrieved (typically the Main Menu or Receptionist Extension)
**Check the box for Use MoH for Ringback so when the caller is parked they hear Music on Hold
**Set Call Limit to 1
**Set the Ring Time to the desired Park duration (up to 180sec)
**Set the Ring Volume to 0 so there is no audible ring at the Park extensions
*[http://wiki.ipitomy.com/wiki/HD_Phones#Manual| Manually Configure] a phone so that Account 1 is 801, Account 2 is 802, Account 3 is 803, etc. Be sure to set each account to use 1 line only. Be sure to choose a phone model that has enough accounts to accommodate however many park destinations the user needs, possibly needing more than one phone if they require a large amount of park locations.
**IP220/290: 2 Accounts
**IP320/330: 3 Accounts
**IP410: 4 Accounts
**IP620: 8 Accounts
*Via the PBX, program BLF keys for 801, 802, 803, on each user Extension that needs to Park calls.

Once this is configured, the user will press a BLF key to Park the call, and the same BLF key to retrieve the call. They must be trained that a Flashing BLF key is in use and therefore cannot be used to park a second call in the same location. Only non-lamped Park destinations are available for use. In essence, we are sending calls to ring the Park Extension. Note: This method will use an extension license for each Directed Park location.

Directed Park

2015-03-13T20:52:52Z

Nick Branica: Added Diredted park feature info

The IPitomy directed park feature allows users to park a call and pick up the parked call on the same key. This is a standard feature in version 5.1.1. There are 20 park "extensions/orbits" 701 - 720. These park keys can be setup as a BLF key as usual. No changes to the park keys need to be made. You may choose to delete the "park" key to free up another button location on the users phone.

It is essential to set BLF transfer mode to "Attended" in the key programming "Display/Call Control" tab on Pone Settings.

The Directed Park feature can be disabled or enabled. If disabled, the standard IPitomy Dynamic parking will work as it always has.

To use the feature, a customer needs to be on a call. While on the call, press the park key/BLF you wish to park the caller on. The caller will be automatically transferred to the park extension/orbit. The BLF assigned to this extension/orbit will blink on all phones that have this programmed as a BLF. Simply press the blinking BLF and pick up the "Parked" call.

The directed park feature can fulfill requirements for users that need departmental parks so only assigned users will see that there is a caller parked in the park orbit. For instance, the receptionist can announce call parked for "parts" and any agent in the parts department will be able to pick up the parked call using the blinking BLF on their phone. Other users can pick up the parked call by dialing the pickup code ((() followed by the specific park orbit i.e. "99701" if 701 is the park orbit assigned to parts.

The feature works in V2 & V3 telephones. It is not possible for this feature to be used in V1 phones. For V1 phones, see below.

'''Alternative parking programming for V1 phones:'''
== Headline text ==

Sometimes an end user is not satisfied with the native Park feature found in the PBX. Instead of Parking calls to the first available orbit, they would rather have the ability to define where the call goes, parking and retrieving the call by pressing the same key and always knowing where its going. In an effort to accommodate this, we have come up with the following solution:

*Build a number of Extensions on the PBX that will be used as Directed Park, in this example we will use 801, 802, 803.
**Set the Busy, NoAnswer, and Unavailable Forwarding for these extensions to a destination that will be the failover if the call is not retrieved (typically the Main Menu or Receptionist Extension)
**Check the box for Use MoH for Ringback so when the caller is parked they hear Music on Hold
**Set Call Limit to 1
**Set the Ring Time to the desired Park duration (up to 180sec)
**Set the Ring Volume to 0 so there is no audible ring at the Park extensions
*[http://wiki.ipitomy.com/wiki/HD_Phones#Manual| Manually Configure] a phone so that Account 1 is 801, Account 2 is 802, Account 3 is 803, etc. Be sure to set each account to use 1 line only. Be sure to choose a phone model that has enough accounts to accommodate however many park destinations the user needs, possibly needing more than one phone if they require a large amount of park locations.
**IP220/290: 2 Accounts
**IP320/330: 3 Accounts
**IP410: 4 Accounts
**IP620: 8 Accounts
*Via the PBX, program BLF keys for 801, 802, 803, on each user Extension that needs to Park calls.

Once this is configured, the user will press a BLF key to Park the call, and the same BLF key to retrieve the call. They must be trained that a Flashing BLF key is in use and therefore cannot be used to park a second call in the same location. Only non-lamped Park destinations are available for use. In essence, we are sending calls to ring the Park Extension. Note: This method will use an extension license for each Directed Park location.

IPitomy dialer app.pdf

2014-04-29T18:23:22Z

Nick Branica: dialer, ipitomy dialer, app,

IPitomy Dialer

[[File:IPitomy_Dialer_App.pdf]]

IPitomy dialer app.pdf

2014-04-29T18:18:52Z

Nick Branica: Replaced content with "File:IPitomy_Dialer_App.pdf"

[[File:IPitomy_Dialer_App.pdf]]

IPitomy dialer app.pdf

2014-04-29T18:16:54Z

Nick Branica: Created page with "IPitomy Dialer Application To install, go to the IPitomy Support Web page and install the one click application installer. Once the software is installed it must be setup. O..."

IPitomy Dialer Application
To install, go to the IPitomy Support Web page and install the one click application installer.

Once the software is installed it must be setup.

Overview
Many users of business phone systems call a wide variety of phone numbers. They can be from a printed sheet, a contact
program or from Microsoft Outlook. When there are many calls to make and each number has to be dialed into a phone,
the process becomes tedious. To ease the process of dialing each number, IPitomy has created the IPitomy Dialer. The
dialer can be used in several ways:
1) Automatically import contacts from Microsoft Outlook. Once the contacts have been imported, simply click on a
contact to dial their number. Your phone will go off hook and the call will be put through without the need to enter
any digits.
2) Import Contacts from a .csv (comma separated values) file. Once the contacts have been imported, simply click
on a contact to dial their number. Your phone will go off hook and the call will be put through without the need to
enter any digits.
3) Dialing System speed dial numbers. The PBX will automatically import any system speed dial numbers. Once
the contacts have been imported, simply click on a contact to dial their number. Your phone will go off hook and the
call will be put through without the need to enter any digits.
4) Click to dial from Mozilla Firefox. using the Telify add-on, phone numbers in web pages can be clicked to dial.
4) Clip to Dial - Copy - by selecting a phone number in any program running on your desktop, Select the phone
number by highlighting it with a mouse. Click Control C then Control D and the number is dialed. Your phone will go
off hook and the call will be put through without the need to enter any digits.
[[File:dialer_settings.png]]

File:Dialer settings.PNG

2014-04-29T18:15:44Z

Nick Branica: IPitomy Dialer screen shot -> Creation failed: Systemcheck showed missing or incorrect configurated, dependent commands!

IPitomy Dialer screen shot

File:IPitomy Dialer App.pdf

2014-04-29T18:07:13Z

Nick Branica: uploaded a new version of "File:IPitomy Dialer App.pdf": newer version

Installation and setup guide for the IPitomy Dialer Application

File:IPitomy Dialer App.pdf

2014-04-29T17:05:03Z

Nick Branica: Installation and setup guide for the IPitomy Dialer Application -> Creation failed: Systemcheck showed missing or incorrect configurated, dependent commands!

Installation and setup guide for the IPitomy Dialer Application

File:IPitomy Communications Credit App.rtf

2013-09-16T21:02:02Z

Nick Branica: Credit Applications for Dealers -> Creation failed: Systemcheck showed missing or incorrect configurated, dependent commands!

Credit Applications for Dealers

Softphones

2013-07-30T15:54:26Z

Nick Branica: /* Chat/Presence with Bria */

= Counterpath Xlite Softphone and the IPitomy PBX =
<div>Download</div>
* Using your web browser, navigate to [http://www.counterpath.com/x-lite.html http://www.counterpath.com/x-lite.html] and
*download the Xlite phone
* Install the Xlite Softphone on your PC
<div>Configuration</div>
* Launch Xlite and you will see the following window

[[File:SoftphonePicture1.png|center]]

*Click Softphone at the top of the phone and select Account Settings 
*Enter the Extension Number for: Display Name, User Name, and Authorization User Name
*Enter the SIP Password of the extension in the Password field
*Enter the appropriate IP address for the PBX under Domain
*Check the box for ‘Register with domain and receive incoming calls
[[File:softphonePicture2.png|center]]
*Click OK and the phone will register to the PBX
<div> </div><div>Congratulations, you have now configured your Xlite softphone to work with the IPitomy PBX.</div><div>You should now be able to see that the phone is registered in the PBX under Monitoring, and you</div><div>will now be able to make calls.</div><div>[Note] In order for DTMF to be correctly recognized, the ext for the Xlite needs to be set to</div><div>rfc2833. Since the free Xlite phone only offers one channel, you will not be able to use the Xfer</div><div>button on the phone itself. In order to transfer you will need to dial ##<extnum>#.</div><div> </div>

= Bria Softphone Setup =

To Successfully set up your Bria Softphone with you IPitomy system, fill out the SIP Account window once you have downloaded the Bria Software.

Your Extension Number which is found under General Settings is used as your User ID followed by the @ symbol and your outside IP address.
<div>[[File:SoftphonePicture3.png]][[File:SoftphonePicture4.png]] </div><div> </div><div>Next, Click on the Advanced Settings link, shown on the bottom of the image above and to the left.</div><div>Once you have reached Advanced Settings, your SIP password will be accessable through the SIP Password generator. You will click generate and once a numer/letter combination appears, insert this password into your password field in the SIP Account window for the Bria Softphone. </div><div> </div><div>[[File:SoftphonePicture5.png]][[File:SoftphonePicture3.png]] </div><div> </div><div> </div><div> </div><div>Congratulations! You have now successfully integrated your IPitomy extension and Bria softphone!</div>

= Chat/Presence with Bria =
By configuring the XMPP client in your Bria soft phone on your cell phone, you can monitor chat and presence of extensions on your PBX.

Step By Step Guide -[[File:Bria_xmpp.pdf]]

*You will need to "add account" and choose XMPP - Messaging
*Then configure that acct.
*AcctName = ext number
*username = ext num
*password = Voicemail PIN
*Domain = locahost
*Then back out to Account Advanced and set the Outbound Proxy to the PBX IP address
*Ensure that port 5022 is forwarded from the Router to the PBX internal IP address

Softphones

2013-07-30T15:53:42Z

Nick Branica: /* Chat/Presence with Bria */

= Counterpath Xlite Softphone and the IPitomy PBX =
<div>Download</div>
* Using your web browser, navigate to [http://www.counterpath.com/x-lite.html http://www.counterpath.com/x-lite.html] and
*download the Xlite phone
* Install the Xlite Softphone on your PC
<div>Configuration</div>
* Launch Xlite and you will see the following window

[[File:SoftphonePicture1.png|center]]

*Click Softphone at the top of the phone and select Account Settings 
*Enter the Extension Number for: Display Name, User Name, and Authorization User Name
*Enter the SIP Password of the extension in the Password field
*Enter the appropriate IP address for the PBX under Domain
*Check the box for ‘Register with domain and receive incoming calls
[[File:softphonePicture2.png|center]]
*Click OK and the phone will register to the PBX
<div> </div><div>Congratulations, you have now configured your Xlite softphone to work with the IPitomy PBX.</div><div>You should now be able to see that the phone is registered in the PBX under Monitoring, and you</div><div>will now be able to make calls.</div><div>[Note] In order for DTMF to be correctly recognized, the ext for the Xlite needs to be set to</div><div>rfc2833. Since the free Xlite phone only offers one channel, you will not be able to use the Xfer</div><div>button on the phone itself. In order to transfer you will need to dial ##<extnum>#.</div><div> </div>

= Bria Softphone Setup =

To Successfully set up your Bria Softphone with you IPitomy system, fill out the SIP Account window once you have downloaded the Bria Software.

Your Extension Number which is found under General Settings is used as your User ID followed by the @ symbol and your outside IP address.
<div>[[File:SoftphonePicture3.png]][[File:SoftphonePicture4.png]] </div><div> </div><div>Next, Click on the Advanced Settings link, shown on the bottom of the image above and to the left.</div><div>Once you have reached Advanced Settings, your SIP password will be accessable through the SIP Password generator. You will click generate and once a numer/letter combination appears, insert this password into your password field in the SIP Account window for the Bria Softphone. </div><div> </div><div>[[File:SoftphonePicture5.png]][[File:SoftphonePicture3.png]] </div><div> </div><div> </div><div> </div><div>Congratulations! You have now successfully integrated your IPitomy extension and Bria softphone!</div>

= Chat/Presence with Bria =
By configuring the XMPP client in your Bria soft phone on your cell phone, you can monitor chat and presence of extensions on your PBX.
[[File:Bria_xmpp.pdf]]

*You will need to "add account" and choose XMPP - Messaging
*Then configure that acct.
*AcctName = ext number
*username = ext num
*password = Voicemail PIN
*Domain = locahost
*Then back out to Account Advanced and set the Outbound Proxy to the PBX IP address
*Ensure that port 5022 is forwarded from the Router to the PBX internal IP address

File:Bria xmpp.pdf

2013-07-30T15:51:13Z

Nick Branica: Step by Step Guide to setup the XMPP client on a Bria Softphone. -> Creation failed: Systemcheck showed missing or incorrect configurated, dependent commands!

Step by Step Guide to setup the XMPP client on a Bria Softphone.

Training: Ring Groups

2013-06-25T13:22:17Z

Nick Branica:

Ring groups are destinations where a single call can be routed so one to many extensions can ring simultaneously. [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg Training Video:] [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg [1]]

IP PBX Manual Destinations Groups

2013-06-25T13:14:38Z

Nick Branica:

{{IP_PBX_Manual|sortkey=Groups}} '''Groups'''

A Group is a logical grouping of extensions and their privileges. Each Group is associated with a set of PBX features and call routing strategies. It is advised to build the extensions in the PBX first, allowing the groups to be populated at the time of creation.

These Groups allow incoming calls to be distributed to a group of extensions rather than just one extension. Within the Groups, different distribution strategies may be selected based on the call coverage required by the application. Groups also provide off-hook paging functionality. By dialing the intercom button or code followed by the group number, extensions in the group will receive the page over the intercom. Groups are a set of extensions that are related either because they:

*Serve a similar business function.
*Work within the same department.
*Are located in proximity to each other.

For example, a business might create a few user Groups in an IP telephony network, e.g. Sales, Marketing, Administration, Accounting, Engineering, etc. Groups may also be created for people in similar locations like a plant floor, the north section of a building or the front office. The goal of a group is to ring telephones based on the incoming routing logic (e.g. DIDs, Auto Attendant selected by the incoming caller, or time of day). Note that calls routing to an extension from a ring group will ignore that phones PBX forwarding settings. Some phones have forwarding settings which are independent of the PBX (not set through the PBX Web GUI). These settings will still be applied.

*Link to [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg '''''Ring''''' '''''Group'''''] training video - [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg [1]]

[[File:Ringgroupspages.png|center|Ringgroupspages.png]]

=== Ring Group Examples ===

Ring groups define a set of extensions (people) that answer calls. Ring groups can be created for departments (e.g. Sales, Engineering) or buisiness regions (e.g. north, south, etc), or areas of a business (e.g. a warehouse or plant floor). These ring groups can appear on an Automated Attendant (menu). When the group is selected from a Menu, the call is routed to the group. These calls will be distributed to the member extensions based upon the ring strategy for the group. The ring strategy for the group can be set from the drop-down list. Available call distribution options are:

*Ring AllRing all extensions in the group simultaneously
*Round RobinTake turns ringing each extension, starts at the first extension for each new call
*Round Robin (with memory)Take turns ringing each extension, starts where the last call left off
*Least RecentRing extensions in order of which was least recently called by the queue. If using agents, logging out will reset the call counter so the next time that agent logs in, they will be at the top of the list for queue calls.
*Fewest CallsRing extension in order of which has the fewest completed calls from the queue
*RandomRandomly ring one extension at a time

==== Example Ring Group 1Departmental Grouping ====

Ring group definitions can be found in the '''Add a New Ring Group''' section of this guide. If a business has the following departments and people: This business can setup the following ring groups supporting business operations.

{| style="border-spacing:0;"
|-
| style="background-color:#4f81bd;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;" | '''GROUP 1 - Sales'''
| style="background-color:#4f81bd;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;" | '''Ext.'''
| style="background-color:#c0504d;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;" | '''GROUP 2Customer Services'''
| style="background-color:#c0504d;border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | '''Ext.'''
|-
| 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;" | Cathy Caldwell
| 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;" | 123
| 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;" | Gretchen Goodall
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 134
|-
| 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;" | David Dawson
| 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;" | 124
| 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;" | Peter Polk
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 135
|-
| 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;" | Susan Smith
| 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;" | 125
| 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;" |
|-
| 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;" | Robert Reed
| 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;" | 126
| 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;" |
|}

Using this ring group scenario the '''Menu would look like''':

*Sales (Destination - Group 1).
*Customer Service (Destination - Group 2).
*Office Manager (Destination - Ext. 113).

The menu prompt (message) for this menu and group arrangement would read as follows:

'''“Thank you for calling CompanyA, a leading manufacturer of cable assemblies and wiring harnesses. If you know the extension of the party you would like to reach you may dial it at any time.”'''

*For Sales, press 1.
*For Customer Service, press 2.
*For Accounting, press 3.

Once a call is sent to Sales, member extension will be rang according to the '''ring group strategy.'''

==== Example Ring Group 2Regional Sales Grouping ====

{| style="border-spacing:0;"
|-
| style="background-color:#4f81bd;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;" | '''GROUP 1 East Coast Sales'''
| style="background-color:#4f81bd;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;" | '''Ext.'''
| style="background-color:#c0504d;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;" | '''GROUP 2 -West Coast Sales'''
| style="background-color:#c0504d;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;" | '''Ext.'''
| style="background-color:#9bbb59;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;" | '''GROUP 3 Customer Service'''
| style="background-color:#9bbb59;border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | '''Ext.'''
|-
| 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;" | Cathy Caldwell
| 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;" | 123
| 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;" | Susan Smith
| 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;" | 125
| 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;" | Gretchen Goodall
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 134
|-
| 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;" | David Dawson
| 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;" | 124
| 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;" | Robert Reed
| 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;" | 126
| 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;" | Peter Polk
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 135
|}

Using this ring group scenario the '''Menu would look like''':

*East Coast Sales (Destination - Group 1).
*West Coast Sales (Destination - Group 2).
*Customer Service (Destination - Group 3).
*Office Manager (Destination - Ext. 113).

The menu '''prompt '''for this menu and group arrangement would read as follows:

'''“Thank you for calling CompanyA, a leading manufacturer of cable assemblies and wiring harnesses. If you know the extension of the party you would like to reach you may dial it at any time. For”'''

*East Coast Sales, press 1.
*West Coast Sales, press 2.
*Customer Service, press 3.
*Accounting, press 4.

Calls sent to these groups might use different ring strategies. The East Coast Sales group might answer calls Round Robin, distributing the calls to each Sale Representative consecutively. If a sales person is missing from the West Coast team this group might set phones to Ring All in the group so that calls don’t get missed. The Customer Service team may get high volumes of calls during a specified period of time, so this group may be set to ring Least Recent.

==== Add/Edit New Ring Group ====

[[File:Editringgroup.png|center|Editringgroup.png]] The following table describes the parameters (fields) on the Edit Ring Group page and their descriptions and recommended settings.

{| border="1" class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| style="text-align: center;" | '''Name''' 
| The name associated with this ring group.
|-
| style="text-align: center;" | ''' Group Number (to dial group)''' 
| The number dialed to access this ring group. Must be 3 or 4 digits.
|-
| style="text-align: center;" | '''Allow Paging with **+Group Number''' 
| If enabled (checked), this will allow <nowiki>**+ </nowiki>group number to page this ring group
|-
| style="text-align: center;" | '''Ring Strategy''' 
| This defines how calls are distributed to member extensions. Options are:
'''Ring All''' - Ring all extensions in the group simultaneously

'''Round Robin''' - Ring extensions consecutively. Delivers a new call to the first person in the group only after the last person in the group has taken a call. If an extension is busy the call will automatically be routed to the next extension in the group.

'''Round Robin (with memory)''' - Ring extensions consecutively. Delivers a new call to the first person in the group only after the last person in the group has taken a call. Remembers where the last call was taken and distributes new calls to the next extension in the rotation.

'''Least Recent''' - Ring extensions in order of which was least recently called by the queue.

'''Fewest Calls''' - Ring extension in order of which has the fewest completed calls from the queue

'''Random''' - Distributes calls randomly to the group.

|-
| style="text-align: center;" | '''Failover Destination''' 
| This defines where a call will be sent when it goes unanswered for a duration that matches the Timeout for the group.
|-
| style="text-align: center;" | '''Timeout''' 
| Defines, in seconds, the total time a call will spend in the ring group before being sent to the Failover Destination
|-
| style="text-align: center;" | '''Allow Recording''' 
| Determines if feature code recording is allowed for calls in this queue.
If YES = enables the recording feature.

If NO = disables the recording feature.

|-
| style="text-align: center;" | '''Agent Ring Time''' 
| Defines, in seconds, how long the queue will ring a particular agent. If using the Ring All strategy, this value should match the Timeout. For other strategies, this should be set to how long an extension should ring before moving on to the next.
|-
| style="text-align: center;" | '''Autofill''' 
|
This deals with how the queue handles multiple concurrent calls. If set to NO the queue will wait until the first call is answered before ringing phones with the second call. If set to YES the queue will ring phones for all calls as they come in.  Note this setting has a different effect with ring all that may not be desirable. 

When autofill is yes and ring strategy is 'ringall' the group will still deliver the calls in the order they are received and in order to prevent calls from being lost by phones that have too few lines for the number of calls in the queue, the group will only deliver a single additional call when Ring In Use is set to yes. If someone who's phone is ringing rejects the call, the group will deliver the next call in line to all members.  This means the the rejected call is no longer ringing the member's phone that rejected the call, however it is still ringing all of the other members.  At the same time the 2nd call in line comes in and rings everyone.

|-
| style="text-align: center;" | '''Ring In use''' 
| If '''YES''' = Distribute calls to group members if they are already on a call
If '''NO''' = Do NOT distribute calls to group members if they are already on a call

|}

The following outlines the steps to add/edit ring group settings.

'''STEPS:'''

#From the '''Destinations'''=>'''Group''' page, click the '''ADD Group''' button. The '''Edit Ring Group''' page appears.
#In the '''Name''' field, enter a name for the group.
#In the '''Group Number''' field, enter a number for the group. This number must be three or four digits in length.
#Check the '''Allow Paging with <nowiki>**+</nowiki> the Group Number''' to enable the setting.
#Select the desired '''Ring Strategy''' from the drop-down list.
#Select the desired '''Failover Destination''' from the list of available options.
#Enter the time for the '''Timeout''' parameter.
#Select “'''Yes'''” if you want to enable the '''Allow Recording '''setting. Otherwise, set it to “'''No'''”.
#Enter the time for the '''Agent Ring Time''' parameter.
#Select “'''Yes'''” if you want to enable the '''Autofill '''setting. Otherwise, set it to “'''No'''”
#Select “'''Yes'''” if you want to enable the '''Ring in use '''setting. Otherwise, set it to “'''No'''”
#Click [[File:Savechanges.png]] the button.
#From the '''Members''' section of the '''Edit Ring Group''' page, select the extension number from the list of extensions that you want to add to the ring group. Click the '''ADD''' button.
#A second list appears next to the Members list displaying the selected extension. Repeat step 13 until all extensions that you want to add are on the list. Additionally, Shift or Control click functionality works for adding multiple extensions at once.
#To remove an item from the list of additions, select the extension number then click the '''REMOVE''' button.
#Define '''Priorities''' for the extensions, if needed. The scale ranges 0 thru 9 (0 is the highest priority).
#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.

==== Ring Group Advanced Settings ====

[[File:Customcallerid.png|center|Customcallerid.png]] The following table describes the parameters (fields) on the '''Advanced''' '''Settings''' for Ring Groups page and their descriptions and recommended settings.

{| class="wikitable"
|-
| colspan="2" | <center>'''Sections/Fields'''</center>
| <center>'''Description'''</center>
|-
| colspan="3" | <center>'''Ring/Hold Settings'''</center>
|-
| style="text-align: center;" | '''Caller Hears'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Override Hold Music'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Hold Music'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Ring Tone'''
| colspan="2" | When Caller Ring Settings are set to Ringing, use this to define the ring cadence.
|-
| colspan="3" | <center>'''Custom Caller ID Settings'''</center>
|-
| style="text-align: center;" | '''Use'''
| colspan="2" | Defines what method, if any, you wish to use to override Inbound CID. The options are:
System Default, Name, Number, Name and Number

|-
| style="text-align: center;" | '''Name'''
| colspan="2" | This is the name of the caller that will be displayed when a call is received.
|-
| style="text-align: center;" | '''Number'''
| colspan="2" | This is the number of the caller that will be displayed when a call is received.
|-
| style="text-align: center;" | '''Prepend'''
| colspan="2" | Selecting Prepend will enter the text you put in the field, followed by the original CID. If not prepending, the override eliminates the original information.
|}

==== Edit Advanced Ring Group Settings ====

'''STEPS:'''

#Select '''Destinations=>Groups'''. The '''Ring Groups''' page appears.
#Select the Ring Group you want to edit by either select the [[File:Penciledit.png]] icon to the right of the group name or click on the blue Group Name link.
#The '''Edit Ring Group''' page appears. Scroll down the page and click on the '''Advanced''' link. The '''Advanced Ring Group Settings''' section is displayed.
#Make the necessary changes to the settings then click button [[File:Savechanges.png]] 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.

==== Edit Custom Caller ID Settings ====

'''STEPS:'''

#Select '''Destinations=>Groups'''. The '''Ring Groups''' page appears.
#Select the Ring Group you want to edit by either select the [[File:Penciledit.png]] icon to the right of the group name or click on the blue Group Name link.
#The '''Edit Ring Group''' page appears. Scroll down the page and click on the '''Advanced''' link. '''The Custom Caller ID''' section is displayed.
#Make the necessary changes to the settings then click [[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.

==== Edit Members/Agents ====

[[File:Editmembersagents.png|center|Editmembersagents.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Members
| This is the list of available extensions that are in the database.
|-
| Agents
| This is the list of available agents that are in the database.
|-
| Priority
| This field allows you to set the priority level for the members or agents. The lower the number, the higher the priority.
|-
| ADD
| This button will add the select item on the list to the right of the available members or agents. Shift and Control click functionality works for adding more than one at a time.
|-
| REMOVE
| This button will remove the item(s) that have been selected from the list of available members or agents. Shift and Control click functionality works for adding more than one at a time.
|}

==== Add Agents/Members to the Group ====

'''STEPS:'''

#Navigate to the '''Destinations=>Groups=>Edit Group''' page.
#Select from the left hand column which agents/members to add to the group.
#Click the '''Add '''button.
#Click [[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 Agents/Members from the Group ====

'''STEPS:'''

#STEPS: Navigate to the '''Destinations'''=>'''Groups'''=>'''Edit Group '''page.
#Select from the right hand column which agents/members to remove from the group.
#Click the '''Add '''button.
#Click [[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

=== Automatic Call Distribution ([[ACD]]) ===

Systems licensed to use [[ACD]] will have access to additional functionality pertaining to Ring Groups. ACD allows for more flexibility and control over the queuing in a Ring Group, as well as allowing the creation of Agents who can be added to the Ring Group.

[[File:Autocalldistpage.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Agent Retry Timer
| This defines, in seconds, how long the queue will wait before attempting to ring an agent again.
|-
| Weight
| The priority of the queue in relation to other queues in the system.
|-
| Wrap-up Time
| This defines, in seconds, how long the queue will wait after an agent has ended a call before flagging then as an active agent ready to accept more calls.
|-
| Autopause
| This will place an agent in a pause state in relation to this queue when they ignore or answer a call. To become an active member of the queue again unpause using the 1* feature code a phone or 1*+agent# for an agent.
|-
| Maximum # of people in queue
| Defines how many callers can join the queue before it is considered full.
|-
| Announce Frequency
| Must be set for Announce Holdtime to function. This pertains to how often the caller will hear the hold time announcement, in seconds
|-
| Periodic Announce Frequency
| This defines how much time passes between playing the recording that was set for this queue under Periodic Announcement.
|-
| Announce Holdtime
| If set to YES this will announce a caller’s status in the queue with regards to their expected hold time, pending the hold time is estimated to be over 1 minute. If the hold time is less than one minute, this will announce the caller’s position in the queue. A period of time is needed for the algorithm to ‘learn’ and estimate hold times accurately. This message will not play for the first caller in the queue. Caller Ring Settings must be set to a MOH file in order for this message to play.
|-
| Join Empty Queue
| Controls if a caller can join a queue.
YES = Callers can join the queue with no members or whose members are busy.

NO = Callers can only join the queue with active, non-busy members.

STRICT = Callers can join the queue as long as there are active members, be they available or busy.

|-
| Leave Empty Queue
| Controls if a caller will leave an empty queue.
YES = Callers will leave the queue if there are no active members, or all members are busy.

NO = Callers will not leave the queue even if there are no active members, or all members are busy.

STRICT = Callers will leave the queue if there are no active members, but will remain in the queue if all members are busy.

|-
| Report Hold Time
| Set to YES if you want the person answering the queue call to hear a message indicating how long the call they just answered was in the queue
|-
| Member Delay before connect
| Delay, in seconds, before a member is connected to the queue.
|-
| Timeout Restart
| If set to YES the ring group timeout will reset after attempting to ring an agent/member. When set to NO, time taken to ring an agent/member will be subtracted from the ring group time, and when the timer ends the group will send the caller to the Failover Destination
|-
| Service Level
| Set this to the time, in seconds, that defines the acceptable level of service for the queue. The Live Queue Data page will track the percentage of calls answered within the allotted time.
|-
| Exit Status: FULL
Exit Status: JOINEMPTY

Exit Status: LEAVEEMPTY

Exit Status: JOINUNAVAIL

Exit Status: LEAVEUNAVAIL

| These are the destinations where calls will go based on the exit strategy defined. If no one in the group accepts the call this is the destination to which the call will be sent. Destinations can be extensions, other ring groups, a menu or a voicemail box.
|-
| Intro Announcement
| Define what prompt to play to the caller just before they enter the queue. (Previously “Announce Filename”)
|-
| Agent Announcement
| Defines what prompt to play to the agent/member who answers the queue call. This setting is set per Queue and therefore can be used to identify what Queue is being answered (for agents handling calls from multiple Queues.
|-
| Periodic Announcement
| Defines what prompt to play to the caller at the Periodic Announce frequency interval. This will not play for the caller in the first position of the queue. Caller Ring Settings must be set to a MOH file in order for this message to play.
|-
| Exit Menu
| Set this to a menu. Key presses while in this group will send people the corresponding destination from that menu.
|}

''Table 39Edit Ring Group ACD Agents Settings and Descriptions''

==== Edit Automatic Call Distribution (ACD) Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Make the necessary updates to the ACD settings.
#Click [[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.

==== Add Automatic Call Distribution (ACD) Agents ====

[[File:Addeditagentfeatures.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Name
| Enter a name that will identify this agent.
|-
| ID
| Enter a unique 3-4 digit number to identify this agent. This number is used to log in/out.
|-
| PIN
| Enter a minimum of 4 digits for the PIN. This is the password used to log in/out with the agent.
|}

''Table 40Add/Edit Agent Features and Descriptions''

 '''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The Agents page appears.
#Click on the '''ADD''' button to add a new agent. A box with the name, ID and PIN field appears. Enter the information for the new agent.
#Click [[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.

==== Edit Automatic Call Distribution (ACD) Agent ====

'''STEPS:'''

#From the '''Destinations''' '''Groups Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The '''Agents''' page appears.
#Update the necessary information for agent(s) then click the '''SAVE CHANGES''' button. The changes that you made will appear in the updated list.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

==== Delete Automatic Call Distribution (ACD) Agent ====

'''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The '''Agents''' page appears.
#From the list of existing agents, click on the box (placing a checkmark) to the left of the Agent(s) that you want to delete. Click the '''DELETE SELECTED''' button.
#The agent selected is removed from the list.
#Click [[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.

=== Multicast Paging Group ===

A Multicast Paging Group provides a way to page a large number of phones, while keeping the burden on the network and PBX to a minimum.

'''NOTE: IPitomy recommends using multicast paging for groups of 10 or more phones that are paged frequently.''' [[File:Multicastgroups.png|center|Multicastgroups.png]] [[File:Editmultipagewindow.png|center|Editmultipagewindow.png]] The following table describes the fields (parameters) on the Multicast Edit Paging Group window and the recommended settings.

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Name
| This is a name of the multicast paging group.
|-
| Local Number
| This is the number that will be dialed to launch the multicast paging to the defined group. This is the unique 3-4 digit number.
|-
| Multicast Address
| This should be a valid multicast IP for a Multicast Address<ref name="ftn1">224.0.0.0 to 239.255.255.255, with many addresses reserved.</ref>. We recommend '''225.5.5.5'''. You can use the same IP for each group, so long as you assign them different ports.
|-
| Multicast Port
| This must be a valid, unused port Multicast Port.
'''NOTE: This must be a port below 6535.'''

|-
| Time to Live
| Time to Live is a counter that decreases by 1 each time a packet passes through a router. When the counter reaches 0, the packet is considered dead and will not pass to any other networks. This can be left blank or configured as needed for your particular install.
'''NOTE: Leaving it blank the packet will not pass to another network.'''

|-
| Members
| The Members section allows you to add the extensions to be paged within the group when a page is sent.
|-
| Add/Remove
| The Add and Remove button allows you to add or remove members (extensions) from the paging group.
|-
| Create/Update
| Depending on the process that is being performed, this button will appear (be labeled) as either Create or Update. This will create a new member or process the changes made.
|-
| Close Window
| This link will close the Edit Page Group window and return you to the Ring Groups page.
|}

The following section outlines the steps to add a new Multicast Paging Group.

==== Add Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate to the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on the '''Add Groups '''button. The '''Edit Page Group''' page appears.
#Enter the necessary information for the new paging group.
#From the '''Members''' section of the '''Edit Page Group''' window, select the members (extensions) that you want to include in the paging group.

Use the CTRL or SHIFT button to select multiple or a range of extensions from the list.

#Click on the '''ADD''' button to add the selected members to the group.
#Click the '''CREATE''' button to save the changes. The “Group Created” message appears. Close the window (click on the “X”). The system returns you to the '''Ring Groups''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones that correspond to extensions that were added to the '''Multicast Paging Group'''. Once all the phones have been re-booted, you will be able to utilize this feature by simply dialing the Local Number that was just created.

==== Edit Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on [[File:Penciledit.png]] icon to the right of the '''Name '''of the paging group you want to update. The '''Edit Page Group''' page appears.
#Enter the necessary information for the paging group.
#Click the '''UPDATE''' button to save the changes. Close the window (click on the “X”). The system returns you to the '''Ring Groups''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones for all the extensions that were involved in the Multicast Paging Group.

==== Delete Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on [[File:Deleteselected.png]] icon to the right of the '''Name '''of the paging group you want to delete. The group is removed from the '''Multicast Paging Group''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones for all the extensions that were involved in the Multicast Paging Group which was deleted.

=== Live Queue Data ===

The Live Queue Data page displays current call statistics for ACD queues. The page will refresh every 10 seconds, allowing a manager to monitor the activity in the queue as it happens. You can Apply Changes if you would like to reset the data.

[[File:Ringlivequedata.png|center|Ringlivequedata.png]]

{| style="border-spacing: 0px;" class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| <queue selection box>
| In the queue selection box the Queue or Queues to be displayed can be selected. Use the Control key (Command on Mac) to select multiple queues for display.
|-
| <queue statistics box>
| The Queue Statistics box has been reformatted in SW release 3.4.1 to allow its use on call center wall boards (large PC displays).
Each queue is encapsulated with a thin-line box. Queue statistics are color-coded for quick orientation on wall boards. Since this data is presented in GUI format it can be easily resized to accommodate large displays.

The top line of the queue display presents the Queue Name and the date.

Statistics are a running total and will reset to zero whenever Apply Changes is clicked (as is the case of the screen capture above).

|-
| Time in Q
| This is the average time that callers have waited in queue before being answered by a queue member/agent.
|-
| Calls in Q
| This is the number of calls that are waiting in the queue.
|-
| Answered
| This is the number of calls answered by members/agent of this queue.
|-
| Abandoned
| This is the number of calls that have not been answered and the caller abandoned the queue.
|-
| Target Time
| Displays the service level set for this ACD queue. The Service Level is the arbitrary time allocated by the queue administrator which is set as a gauge by which all calls should be answered.
|-
|  % in Target
| This is the percentage of answered calls measured against the service level. (The percentage of calls being answered within the set Target Time.)
|-
| <Member/Agent field>
| At the right side of the Queue Statistics box the agents and queue members are listed. There are two maximum columns and the window will resize to accommodate all members and agents registered to this queue.
Member/Agent status is available at a glance with color-codes as follows:

{| style="border-spacing:0;"
|-
| style="background-color:#c00000;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Red'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | In Use
|-
| style="background-color:#006600;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Green'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Idle (Logged In)
|-
| style="background-color:#ffff00;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Yellow'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Ringing
|-
| style="background-color:#7f7f7f;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Gray'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Unavailable (Not Logged In)
|-
| style="background-color:#e36c0a;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Orange'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Paused
|}

Also in the Member/Agent data are two numeral fields. The first number represents the Priority of this Member/Agent in this queue. The second number is the total calls taken by this Member/Agent.

|}

=== Live Queue Data Wallboard ===

Software version 3.4.1 and above allows the use of call center wall boards with Live Call Queue. This function is facilitated using a special user name and password via Smart Personal Console.

==== Activate Live Call Queue Wallboard ====

'''STEPS:'''

#Navigate to the '''PBX Setup'''=>'''General '''page
#Locate to the '''General Settings''' section (second section from the top).
#Locate the SPC Reports User Password field (bottom of section)

[[File:Spcreportsuserpass.png]]

#It may be helpful to change this to a user-friendly password. If you change the password be sure to [[File:Savechanges.png]] and '''Apply Changes'''.
#Make a note of this password.
#On the PC that will be used for the wallboard display, open a web browser and navigate to the IPitomy PBX login screen
#Using the right sideSmart Personal Console (User Login) input the word “reports” as the user name and the password saved in step 3. [[File:Reportslogin.png]]
#Click on '''Login''' .
#The CDR-Report page is opened. On this page is a button to select Live Queue.
#Click on '''Live Queue'''.
#The same display presented in PBX Administration is displayed. Simply select the group or groups that you wish to display on the wallboard.
#Size this web page data output for the monitor in use by pressing and holding the Cntrl key (PC, Command key on Mac) and press “+” or “-“ until the data is sized as you wish.

----

<references/>

IP PBX Manual Destinations Groups

2013-06-25T12:34:52Z

Nick Branica:

{{IP_PBX_Manual|sortkey=Groups}} '''Groups'''

A Group is a logical grouping of extensions and their privileges. Each Group is associated with a set of PBX features and call routing strategies. It is advised to build the extensions in the PBX first, allowing the groups to be populated at the time of creation.

These Groups allow incoming calls to be distributed to a group of extensions rather than just one extension. Within the Groups, different distribution strategies may be selected based on the call coverage required by the application. Groups also provide off-hook paging functionality. By dialing the intercom button or code followed by the group number, extensions in the group will receive the page over the intercom. Groups are a set of extensions that are related either because they:

*Serve a similar business function.
*Work within the same department.
*Are located in proximity to each other.

For example, a business might create a few user Groups in an IP telephony network, e.g. Sales, Marketing, Administration, Accounting, Engineering, etc. Groups may also be created for people in similar locations like a plant floor, the north section of a building or the front office. The goal of a group is to ring telephones based on the incoming routing logic (e.g. DIDs, Auto Attendant selected by the incoming caller, or time of day). Note that calls routing to an extension from a ring group will ignore that phones PBX forwarding settings. Some phones have forwarding settings which are independent of the PBX (not set through the PBX Web GUI). These settings will still be applied.

*Link to '''''Ring''''' '''''Group''''' training video - [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg [1]]

[[File:Ringgroupspages.png|center|Ringgroupspages.png]]

=== Ring Group Examples ===

Ring groups define a set of extensions (people) that answer calls. Ring groups can be created for departments (e.g. Sales, Engineering) or buisiness regions (e.g. north, south, etc), or areas of a business (e.g. a warehouse or plant floor). These ring groups can appear on an Automated Attendant (menu). When the group is selected from a Menu, the call is routed to the group. These calls will be distributed to the member extensions based upon the ring strategy for the group. The ring strategy for the group can be set from the drop-down list. Available call distribution options are:

*Ring AllRing all extensions in the group simultaneously
*Round RobinTake turns ringing each extension, starts at the first extension for each new call
*Round Robin (with memory)Take turns ringing each extension, starts where the last call left off
*Least RecentRing extensions in order of which was least recently called by the queue. If using agents, logging out will reset the call counter so the next time that agent logs in, they will be at the top of the list for queue calls.
*Fewest CallsRing extension in order of which has the fewest completed calls from the queue
*RandomRandomly ring one extension at a time

==== Example Ring Group 1Departmental Grouping ====

Ring group definitions can be found in the '''Add a New Ring Group''' section of this guide. If a business has the following departments and people: This business can setup the following ring groups supporting business operations.

{| style="border-spacing:0;"
|-
| style="background-color:#4f81bd;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;" | '''GROUP 1 - Sales'''
| style="background-color:#4f81bd;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;" | '''Ext.'''
| style="background-color:#c0504d;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;" | '''GROUP 2Customer Services'''
| style="background-color:#c0504d;border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | '''Ext.'''
|-
| 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;" | Cathy Caldwell
| 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;" | 123
| 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;" | Gretchen Goodall
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 134
|-
| 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;" | David Dawson
| 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;" | 124
| 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;" | Peter Polk
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 135
|-
| 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;" | Susan Smith
| 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;" | 125
| 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;" |
|-
| 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;" | Robert Reed
| 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;" | 126
| 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;" |
|}

Using this ring group scenario the '''Menu would look like''':

*Sales (Destination - Group 1).
*Customer Service (Destination - Group 2).
*Office Manager (Destination - Ext. 113).

The menu prompt (message) for this menu and group arrangement would read as follows:

'''“Thank you for calling CompanyA, a leading manufacturer of cable assemblies and wiring harnesses. If you know the extension of the party you would like to reach you may dial it at any time.”'''

*For Sales, press 1.
*For Customer Service, press 2.
*For Accounting, press 3.

Once a call is sent to Sales, member extension will be rang according to the '''ring group strategy.'''

==== Example Ring Group 2Regional Sales Grouping ====

{| style="border-spacing:0;"
|-
| style="background-color:#4f81bd;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;" | '''GROUP 1 East Coast Sales'''
| style="background-color:#4f81bd;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;" | '''Ext.'''
| style="background-color:#c0504d;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;" | '''GROUP 2 -West Coast Sales'''
| style="background-color:#c0504d;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;" | '''Ext.'''
| style="background-color:#9bbb59;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;" | '''GROUP 3 Customer Service'''
| style="background-color:#9bbb59;border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | '''Ext.'''
|-
| 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;" | Cathy Caldwell
| 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;" | 123
| 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;" | Susan Smith
| 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;" | 125
| 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;" | Gretchen Goodall
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 134
|-
| 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;" | David Dawson
| 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;" | 124
| 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;" | Robert Reed
| 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;" | 126
| 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;" | Peter Polk
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 135
|}

Using this ring group scenario the '''Menu would look like''':

*East Coast Sales (Destination - Group 1).
*West Coast Sales (Destination - Group 2).
*Customer Service (Destination - Group 3).
*Office Manager (Destination - Ext. 113).

The menu '''prompt '''for this menu and group arrangement would read as follows:

'''“Thank you for calling CompanyA, a leading manufacturer of cable assemblies and wiring harnesses. If you know the extension of the party you would like to reach you may dial it at any time. For”'''

*East Coast Sales, press 1.
*West Coast Sales, press 2.
*Customer Service, press 3.
*Accounting, press 4.

Calls sent to these groups might use different ring strategies. The East Coast Sales group might answer calls Round Robin, distributing the calls to each Sale Representative consecutively. If a sales person is missing from the West Coast team this group might set phones to Ring All in the group so that calls don’t get missed. The Customer Service team may get high volumes of calls during a specified period of time, so this group may be set to ring Least Recent.

==== Add/Edit New Ring Group ====

[[File:Editringgroup.png|center|Editringgroup.png]] The following table describes the parameters (fields) on the Edit Ring Group page and their descriptions and recommended settings.

{| border="1" class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| style="text-align: center;" | '''Name''' 
| The name associated with this ring group.
|-
| style="text-align: center;" | ''' Group Number (to dial group)''' 
| The number dialed to access this ring group. Must be 3 or 4 digits.
|-
| style="text-align: center;" | '''Allow Paging with **+Group Number''' 
| If enabled (checked), this will allow <nowiki>**+ </nowiki>group number to page this ring group
|-
| style="text-align: center;" | '''Ring Strategy''' 
| This defines how calls are distributed to member extensions. Options are:
'''Ring All''' - Ring all extensions in the group simultaneously

'''Round Robin''' - Ring extensions consecutively. Delivers a new call to the first person in the group only after the last person in the group has taken a call. If an extension is busy the call will automatically be routed to the next extension in the group.

'''Round Robin (with memory)''' - Ring extensions consecutively. Delivers a new call to the first person in the group only after the last person in the group has taken a call. Remembers where the last call was taken and distributes new calls to the next extension in the rotation.

'''Least Recent''' - Ring extensions in order of which was least recently called by the queue.

'''Fewest Calls''' - Ring extension in order of which has the fewest completed calls from the queue

'''Random''' - Distributes calls randomly to the group.

|-
| style="text-align: center;" | '''Failover Destination''' 
| This defines where a call will be sent when it goes unanswered for a duration that matches the Timeout for the group.
|-
| style="text-align: center;" | '''Timeout''' 
| Defines, in seconds, the total time a call will spend in the ring group before being sent to the Failover Destination
|-
| style="text-align: center;" | '''Allow Recording''' 
| Determines if feature code recording is allowed for calls in this queue.
If YES = enables the recording feature.

If NO = disables the recording feature.

|-
| style="text-align: center;" | '''Agent Ring Time''' 
| Defines, in seconds, how long the queue will ring a particular agent. If using the Ring All strategy, this value should match the Timeout. For other strategies, this should be set to how long an extension should ring before moving on to the next.
|-
| style="text-align: center;" | '''Autofill''' 
|
This deals with how the queue handles multiple concurrent calls. If set to NO the queue will wait until the first call is answered before ringing phones with the second call. If set to YES the queue will ring phones for all calls as they come in.  Note this setting has a different effect with ring all that may not be desirable. 

When autofill is yes and ring strategy is 'ringall' the group will still deliver the calls in the order they are received and in order to prevent calls from being lost by phones that have too few lines for the number of calls in the queue, the group will only deliver a single additional call when Ring In Use is set to yes. If someone who's phone is ringing rejects the call, the group will deliver the next call in line to all members.  This means the the rejected call is no longer ringing the member's phone that rejected the call, however it is still ringing all of the other members.  At the same time the 2nd call in line comes in and rings everyone.

|-
| style="text-align: center;" | '''Ring In use''' 
| If '''YES''' = Distribute calls to group members if they are already on a call
If '''NO''' = Do NOT distribute calls to group members if they are already on a call

|}

The following outlines the steps to add/edit ring group settings.

'''STEPS:'''

#From the '''Destinations'''=>'''Group''' page, click the '''ADD Group''' button. The '''Edit Ring Group''' page appears.
#In the '''Name''' field, enter a name for the group.
#In the '''Group Number''' field, enter a number for the group. This number must be three or four digits in length.
#Check the '''Allow Paging with <nowiki>**+</nowiki> the Group Number''' to enable the setting.
#Select the desired '''Ring Strategy''' from the drop-down list.
#Select the desired '''Failover Destination''' from the list of available options.
#Enter the time for the '''Timeout''' parameter.
#Select “'''Yes'''” if you want to enable the '''Allow Recording '''setting. Otherwise, set it to “'''No'''”.
#Enter the time for the '''Agent Ring Time''' parameter.
#Select “'''Yes'''” if you want to enable the '''Autofill '''setting. Otherwise, set it to “'''No'''”
#Select “'''Yes'''” if you want to enable the '''Ring in use '''setting. Otherwise, set it to “'''No'''”
#Click [[File:Savechanges.png]] the button.
#From the '''Members''' section of the '''Edit Ring Group''' page, select the extension number from the list of extensions that you want to add to the ring group. Click the '''ADD''' button.
#A second list appears next to the Members list displaying the selected extension. Repeat step 13 until all extensions that you want to add are on the list. Additionally, Shift or Control click functionality works for adding multiple extensions at once.
#To remove an item from the list of additions, select the extension number then click the '''REMOVE''' button.
#Define '''Priorities''' for the extensions, if needed. The scale ranges 0 thru 9 (0 is the highest priority).
#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.

==== Ring Group Advanced Settings ====

[[File:Customcallerid.png|center|Customcallerid.png]] The following table describes the parameters (fields) on the '''Advanced''' '''Settings''' for Ring Groups page and their descriptions and recommended settings.

{| class="wikitable"
|-
| colspan="2" | <center>'''Sections/Fields'''</center>
| <center>'''Description'''</center>
|-
| colspan="3" | <center>'''Ring/Hold Settings'''</center>
|-
| style="text-align: center;" | '''Caller Hears'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Override Hold Music'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Hold Music'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Ring Tone'''
| colspan="2" | When Caller Ring Settings are set to Ringing, use this to define the ring cadence.
|-
| colspan="3" | <center>'''Custom Caller ID Settings'''</center>
|-
| style="text-align: center;" | '''Use'''
| colspan="2" | Defines what method, if any, you wish to use to override Inbound CID. The options are:
System Default, Name, Number, Name and Number

|-
| style="text-align: center;" | '''Name'''
| colspan="2" | This is the name of the caller that will be displayed when a call is received.
|-
| style="text-align: center;" | '''Number'''
| colspan="2" | This is the number of the caller that will be displayed when a call is received.
|-
| style="text-align: center;" | '''Prepend'''
| colspan="2" | Selecting Prepend will enter the text you put in the field, followed by the original CID. If not prepending, the override eliminates the original information.
|}

==== Edit Advanced Ring Group Settings ====

'''STEPS:'''

#Select '''Destinations=>Groups'''. The '''Ring Groups''' page appears.
#Select the Ring Group you want to edit by either select the [[File:Penciledit.png]] icon to the right of the group name or click on the blue Group Name link.
#The '''Edit Ring Group''' page appears. Scroll down the page and click on the '''Advanced''' link. The '''Advanced Ring Group Settings''' section is displayed.
#Make the necessary changes to the settings then click button [[File:Savechanges.png]] 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.

==== Edit Custom Caller ID Settings ====

'''STEPS:'''

#Select '''Destinations=>Groups'''. The '''Ring Groups''' page appears.
#Select the Ring Group you want to edit by either select the [[File:Penciledit.png]] icon to the right of the group name or click on the blue Group Name link.
#The '''Edit Ring Group''' page appears. Scroll down the page and click on the '''Advanced''' link. '''The Custom Caller ID''' section is displayed.
#Make the necessary changes to the settings then click [[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.

==== Edit Members/Agents ====

[[File:Editmembersagents.png|center|Editmembersagents.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Members
| This is the list of available extensions that are in the database.
|-
| Agents
| This is the list of available agents that are in the database.
|-
| Priority
| This field allows you to set the priority level for the members or agents. The lower the number, the higher the priority.
|-
| ADD
| This button will add the select item on the list to the right of the available members or agents. Shift and Control click functionality works for adding more than one at a time.
|-
| REMOVE
| This button will remove the item(s) that have been selected from the list of available members or agents. Shift and Control click functionality works for adding more than one at a time.
|}

==== Add Agents/Members to the Group ====

'''STEPS:'''

#Navigate to the '''Destinations=>Groups=>Edit Group''' page.
#Select from the left hand column which agents/members to add to the group.
#Click the '''Add '''button.
#Click [[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 Agents/Members from the Group ====

'''STEPS:'''

#STEPS: Navigate to the '''Destinations'''=>'''Groups'''=>'''Edit Group '''page.
#Select from the right hand column which agents/members to remove from the group.
#Click the '''Add '''button.
#Click [[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

=== Automatic Call Distribution ([[ACD]]) ===

Systems licensed to use [[ACD]] will have access to additional functionality pertaining to Ring Groups. ACD allows for more flexibility and control over the queuing in a Ring Group, as well as allowing the creation of Agents who can be added to the Ring Group.

[[File:Autocalldistpage.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Agent Retry Timer
| This defines, in seconds, how long the queue will wait before attempting to ring an agent again.
|-
| Weight
| The priority of the queue in relation to other queues in the system.
|-
| Wrap-up Time
| This defines, in seconds, how long the queue will wait after an agent has ended a call before flagging then as an active agent ready to accept more calls.
|-
| Autopause
| This will place an agent in a pause state in relation to this queue when they ignore or answer a call. To become an active member of the queue again unpause using the 1* feature code a phone or 1*+agent# for an agent.
|-
| Maximum # of people in queue
| Defines how many callers can join the queue before it is considered full.
|-
| Announce Frequency
| Must be set for Announce Holdtime to function. This pertains to how often the caller will hear the hold time announcement, in seconds
|-
| Periodic Announce Frequency
| This defines how much time passes between playing the recording that was set for this queue under Periodic Announcement.
|-
| Announce Holdtime
| If set to YES this will announce a caller’s status in the queue with regards to their expected hold time, pending the hold time is estimated to be over 1 minute. If the hold time is less than one minute, this will announce the caller’s position in the queue. A period of time is needed for the algorithm to ‘learn’ and estimate hold times accurately. This message will not play for the first caller in the queue. Caller Ring Settings must be set to a MOH file in order for this message to play.
|-
| Join Empty Queue
| Controls if a caller can join a queue.
YES = Callers can join the queue with no members or whose members are busy.

NO = Callers can only join the queue with active, non-busy members.

STRICT = Callers can join the queue as long as there are active members, be they available or busy.

|-
| Leave Empty Queue
| Controls if a caller will leave an empty queue.
YES = Callers will leave the queue if there are no active members, or all members are busy.

NO = Callers will not leave the queue even if there are no active members, or all members are busy.

STRICT = Callers will leave the queue if there are no active members, but will remain in the queue if all members are busy.

|-
| Report Hold Time
| Set to YES if you want the person answering the queue call to hear a message indicating how long the call they just answered was in the queue
|-
| Member Delay before connect
| Delay, in seconds, before a member is connected to the queue.
|-
| Timeout Restart
| If set to YES the ring group timeout will reset after attempting to ring an agent/member. When set to NO, time taken to ring an agent/member will be subtracted from the ring group time, and when the timer ends the group will send the caller to the Failover Destination
|-
| Service Level
| Set this to the time, in seconds, that defines the acceptable level of service for the queue. The Live Queue Data page will track the percentage of calls answered within the allotted time.
|-
| Exit Status: FULL
Exit Status: JOINEMPTY

Exit Status: LEAVEEMPTY

Exit Status: JOINUNAVAIL

Exit Status: LEAVEUNAVAIL

| These are the destinations where calls will go based on the exit strategy defined. If no one in the group accepts the call this is the destination to which the call will be sent. Destinations can be extensions, other ring groups, a menu or a voicemail box.
|-
| Intro Announcement
| Define what prompt to play to the caller just before they enter the queue. (Previously “Announce Filename”)
|-
| Agent Announcement
| Defines what prompt to play to the agent/member who answers the queue call. This setting is set per Queue and therefore can be used to identify what Queue is being answered (for agents handling calls from multiple Queues.
|-
| Periodic Announcement
| Defines what prompt to play to the caller at the Periodic Announce frequency interval. This will not play for the caller in the first position of the queue. Caller Ring Settings must be set to a MOH file in order for this message to play.
|-
| Exit Menu
| Set this to a menu. Key presses while in this group will send people the corresponding destination from that menu.
|}

''Table 39Edit Ring Group ACD Agents Settings and Descriptions''

==== Edit Automatic Call Distribution (ACD) Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Make the necessary updates to the ACD settings.
#Click [[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.

==== Add Automatic Call Distribution (ACD) Agents ====

[[File:Addeditagentfeatures.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Name
| Enter a name that will identify this agent.
|-
| ID
| Enter a unique 3-4 digit number to identify this agent. This number is used to log in/out.
|-
| PIN
| Enter a minimum of 4 digits for the PIN. This is the password used to log in/out with the agent.
|}

''Table 40Add/Edit Agent Features and Descriptions''

 '''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The Agents page appears.
#Click on the '''ADD''' button to add a new agent. A box with the name, ID and PIN field appears. Enter the information for the new agent.
#Click [[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.

==== Edit Automatic Call Distribution (ACD) Agent ====

'''STEPS:'''

#From the '''Destinations''' '''Groups Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The '''Agents''' page appears.
#Update the necessary information for agent(s) then click the '''SAVE CHANGES''' button. The changes that you made will appear in the updated list.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

==== Delete Automatic Call Distribution (ACD) Agent ====

'''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The '''Agents''' page appears.
#From the list of existing agents, click on the box (placing a checkmark) to the left of the Agent(s) that you want to delete. Click the '''DELETE SELECTED''' button.
#The agent selected is removed from the list.
#Click [[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.

=== Multicast Paging Group ===

A Multicast Paging Group provides a way to page a large number of phones, while keeping the burden on the network and PBX to a minimum.

'''NOTE: IPitomy recommends using multicast paging for groups of 10 or more phones that are paged frequently.''' [[File:Multicastgroups.png|center|Multicastgroups.png]] [[File:Editmultipagewindow.png|center|Editmultipagewindow.png]] The following table describes the fields (parameters) on the Multicast Edit Paging Group window and the recommended settings.

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Name
| This is a name of the multicast paging group.
|-
| Local Number
| This is the number that will be dialed to launch the multicast paging to the defined group. This is the unique 3-4 digit number.
|-
| Multicast Address
| This should be a valid multicast IP for a Multicast Address<ref name="ftn1">224.0.0.0 to 239.255.255.255, with many addresses reserved.</ref>. We recommend '''225.5.5.5'''. You can use the same IP for each group, so long as you assign them different ports.
|-
| Multicast Port
| This must be a valid, unused port Multicast Port.
'''NOTE: This must be a port below 6535.'''

|-
| Time to Live
| Time to Live is a counter that decreases by 1 each time a packet passes through a router. When the counter reaches 0, the packet is considered dead and will not pass to any other networks. This can be left blank or configured as needed for your particular install.
'''NOTE: Leaving it blank the packet will not pass to another network.'''

|-
| Members
| The Members section allows you to add the extensions to be paged within the group when a page is sent.
|-
| Add/Remove
| The Add and Remove button allows you to add or remove members (extensions) from the paging group.
|-
| Create/Update
| Depending on the process that is being performed, this button will appear (be labeled) as either Create or Update. This will create a new member or process the changes made.
|-
| Close Window
| This link will close the Edit Page Group window and return you to the Ring Groups page.
|}

The following section outlines the steps to add a new Multicast Paging Group.

==== Add Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate to the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on the '''Add Groups '''button. The '''Edit Page Group''' page appears.
#Enter the necessary information for the new paging group.
#From the '''Members''' section of the '''Edit Page Group''' window, select the members (extensions) that you want to include in the paging group.

Use the CTRL or SHIFT button to select multiple or a range of extensions from the list.

#Click on the '''ADD''' button to add the selected members to the group.
#Click the '''CREATE''' button to save the changes. The “Group Created” message appears. Close the window (click on the “X”). The system returns you to the '''Ring Groups''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones that correspond to extensions that were added to the '''Multicast Paging Group'''. Once all the phones have been re-booted, you will be able to utilize this feature by simply dialing the Local Number that was just created.

==== Edit Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on [[File:Penciledit.png]] icon to the right of the '''Name '''of the paging group you want to update. The '''Edit Page Group''' page appears.
#Enter the necessary information for the paging group.
#Click the '''UPDATE''' button to save the changes. Close the window (click on the “X”). The system returns you to the '''Ring Groups''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones for all the extensions that were involved in the Multicast Paging Group.

==== Delete Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on [[File:Deleteselected.png]] icon to the right of the '''Name '''of the paging group you want to delete. The group is removed from the '''Multicast Paging Group''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones for all the extensions that were involved in the Multicast Paging Group which was deleted.

=== Live Queue Data ===

The Live Queue Data page displays current call statistics for ACD queues. The page will refresh every 10 seconds, allowing a manager to monitor the activity in the queue as it happens. You can Apply Changes if you would like to reset the data.

[[File:Ringlivequedata.png|center|Ringlivequedata.png]]

{| style="border-spacing: 0px;" class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| <queue selection box>
| In the queue selection box the Queue or Queues to be displayed can be selected. Use the Control key (Command on Mac) to select multiple queues for display.
|-
| <queue statistics box>
| The Queue Statistics box has been reformatted in SW release 3.4.1 to allow its use on call center wall boards (large PC displays).
Each queue is encapsulated with a thin-line box. Queue statistics are color-coded for quick orientation on wall boards. Since this data is presented in GUI format it can be easily resized to accommodate large displays.

The top line of the queue display presents the Queue Name and the date.

Statistics are a running total and will reset to zero whenever Apply Changes is clicked (as is the case of the screen capture above).

|-
| Time in Q
| This is the average time that callers have waited in queue before being answered by a queue member/agent.
|-
| Calls in Q
| This is the number of calls that are waiting in the queue.
|-
| Answered
| This is the number of calls answered by members/agent of this queue.
|-
| Abandoned
| This is the number of calls that have not been answered and the caller abandoned the queue.
|-
| Target Time
| Displays the service level set for this ACD queue. The Service Level is the arbitrary time allocated by the queue administrator which is set as a gauge by which all calls should be answered.
|-
|  % in Target
| This is the percentage of answered calls measured against the service level. (The percentage of calls being answered within the set Target Time.)
|-
| <Member/Agent field>
| At the right side of the Queue Statistics box the agents and queue members are listed. There are two maximum columns and the window will resize to accommodate all members and agents registered to this queue.
Member/Agent status is available at a glance with color-codes as follows:

{| style="border-spacing:0;"
|-
| style="background-color:#c00000;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Red'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | In Use
|-
| style="background-color:#006600;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Green'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Idle (Logged In)
|-
| style="background-color:#ffff00;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Yellow'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Ringing
|-
| style="background-color:#7f7f7f;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Gray'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Unavailable (Not Logged In)
|-
| style="background-color:#e36c0a;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Orange'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Paused
|}

Also in the Member/Agent data are two numeral fields. The first number represents the Priority of this Member/Agent in this queue. The second number is the total calls taken by this Member/Agent.

|}

=== Live Queue Data Wallboard ===

Software version 3.4.1 and above allows the use of call center wall boards with Live Call Queue. This function is facilitated using a special user name and password via Smart Personal Console.

==== Activate Live Call Queue Wallboard ====

'''STEPS:'''

#Navigate to the '''PBX Setup'''=>'''General '''page
#Locate to the '''General Settings''' section (second section from the top).
#Locate the SPC Reports User Password field (bottom of section)

[[File:Spcreportsuserpass.png]]

#It may be helpful to change this to a user-friendly password. If you change the password be sure to [[File:Savechanges.png]] and '''Apply Changes'''.
#Make a note of this password.
#On the PC that will be used for the wallboard display, open a web browser and navigate to the IPitomy PBX login screen
#Using the right sideSmart Personal Console (User Login) input the word “reports” as the user name and the password saved in step 3. [[File:Reportslogin.png]]
#Click on '''Login''' .
#The CDR-Report page is opened. On this page is a button to select Live Queue.
#Click on '''Live Queue'''.
#The same display presented in PBX Administration is displayed. Simply select the group or groups that you wish to display on the wallboard.
#Size this web page data output for the monitor in use by pressing and holding the Cntrl key (PC, Command key on Mac) and press “+” or “-“ until the data is sized as you wish.

----

<references/>

IP PBX Manual Destinations Groups

2013-06-25T12:33:19Z

Nick Branica:

{{IP_PBX_Manual|sortkey=Groups}} '''Groups'''

A Group is a logical grouping of extensions and their privileges. Each Group is associated with a set of PBX features and call routing strategies. It is advised to build the extensions in the PBX first, allowing the groups to be populated at the time of creation.

These Groups allow incoming calls to be distributed to a group of extensions rather than just one extension. Within the Groups, different distribution strategies may be selected based on the call coverage required by the application. Groups also provide off-hook paging functionality. By dialing the intercom button or code followed by the group number, extensions in the group will receive the page over the intercom. Groups are a set of extensions that are related either because they:

*Serve a similar business function.
*Work within the same department.
*Are located in proximity to each other.

For example, a business might create a few user Groups in an IP telephony network, e.g. Sales, Marketing, Administration, Accounting, Engineering, etc. Groups may also be created for people in similar locations like a plant floor, the north section of a building or the front office. The goal of a group is to ring telephones based on the incoming routing logic (e.g. DIDs, Auto Attendant selected by the incoming caller, or time of day). Note that calls routing to an extension from a ring group will ignore that phones PBX forwarding settings. Some phones have forwarding settings which are independent of the PBX (not set through the PBX Web GUI). These settings will still be applied.

*Link to '''''Group''''' training video - [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg [1]]

[[File:Ringgroupspages.png|center|Ringgroupspages.png]]

=== Ring Group Examples ===

Ring groups define a set of extensions (people) that answer calls. Ring groups can be created for departments (e.g. Sales, Engineering) or buisiness regions (e.g. north, south, etc), or areas of a business (e.g. a warehouse or plant floor). These ring groups can appear on an Automated Attendant (menu). When the group is selected from a Menu, the call is routed to the group. These calls will be distributed to the member extensions based upon the ring strategy for the group. The ring strategy for the group can be set from the drop-down list. Available call distribution options are:

*Ring AllRing all extensions in the group simultaneously
*Round RobinTake turns ringing each extension, starts at the first extension for each new call
*Round Robin (with memory)Take turns ringing each extension, starts where the last call left off
*Least RecentRing extensions in order of which was least recently called by the queue. If using agents, logging out will reset the call counter so the next time that agent logs in, they will be at the top of the list for queue calls.
*Fewest CallsRing extension in order of which has the fewest completed calls from the queue
*RandomRandomly ring one extension at a time

==== Example Ring Group 1Departmental Grouping ====

Ring group definitions can be found in the '''Add a New Ring Group''' section of this guide. If a business has the following departments and people: This business can setup the following ring groups supporting business operations.

{| style="border-spacing:0;"
|-
| style="background-color:#4f81bd;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;" | '''GROUP 1 - Sales'''
| style="background-color:#4f81bd;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;" | '''Ext.'''
| style="background-color:#c0504d;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;" | '''GROUP 2Customer Services'''
| style="background-color:#c0504d;border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | '''Ext.'''
|-
| 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;" | Cathy Caldwell
| 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;" | 123
| 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;" | Gretchen Goodall
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 134
|-
| 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;" | David Dawson
| 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;" | 124
| 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;" | Peter Polk
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 135
|-
| 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;" | Susan Smith
| 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;" | 125
| 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;" |
|-
| 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;" | Robert Reed
| 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;" | 126
| 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;" |
|}

Using this ring group scenario the '''Menu would look like''':

*Sales (Destination - Group 1).
*Customer Service (Destination - Group 2).
*Office Manager (Destination - Ext. 113).

The menu prompt (message) for this menu and group arrangement would read as follows:

'''“Thank you for calling CompanyA, a leading manufacturer of cable assemblies and wiring harnesses. If you know the extension of the party you would like to reach you may dial it at any time.”'''

*For Sales, press 1.
*For Customer Service, press 2.
*For Accounting, press 3.

Once a call is sent to Sales, member extension will be rang according to the '''ring group strategy.'''

==== Example Ring Group 2Regional Sales Grouping ====

{| style="border-spacing:0;"
|-
| style="background-color:#4f81bd;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;" | '''GROUP 1 East Coast Sales'''
| style="background-color:#4f81bd;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;" | '''Ext.'''
| style="background-color:#c0504d;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;" | '''GROUP 2 -West Coast Sales'''
| style="background-color:#c0504d;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;" | '''Ext.'''
| style="background-color:#9bbb59;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;" | '''GROUP 3 Customer Service'''
| style="background-color:#9bbb59;border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | '''Ext.'''
|-
| 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;" | Cathy Caldwell
| 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;" | 123
| 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;" | Susan Smith
| 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;" | 125
| 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;" | Gretchen Goodall
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 134
|-
| 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;" | David Dawson
| 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;" | 124
| 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;" | Robert Reed
| 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;" | 126
| 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;" | Peter Polk
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | 135
|}

Using this ring group scenario the '''Menu would look like''':

*East Coast Sales (Destination - Group 1).
*West Coast Sales (Destination - Group 2).
*Customer Service (Destination - Group 3).
*Office Manager (Destination - Ext. 113).

The menu '''prompt '''for this menu and group arrangement would read as follows:

'''“Thank you for calling CompanyA, a leading manufacturer of cable assemblies and wiring harnesses. If you know the extension of the party you would like to reach you may dial it at any time. For”'''

*East Coast Sales, press 1.
*West Coast Sales, press 2.
*Customer Service, press 3.
*Accounting, press 4.

Calls sent to these groups might use different ring strategies. The East Coast Sales group might answer calls Round Robin, distributing the calls to each Sale Representative consecutively. If a sales person is missing from the West Coast team this group might set phones to Ring All in the group so that calls don’t get missed. The Customer Service team may get high volumes of calls during a specified period of time, so this group may be set to ring Least Recent.

==== Add/Edit New Ring Group ====

[[File:Editringgroup.png|center|Editringgroup.png]] The following table describes the parameters (fields) on the Edit Ring Group page and their descriptions and recommended settings.

{| border="1" class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| style="text-align: center;" | '''Name''' 
| The name associated with this ring group.
|-
| style="text-align: center;" | ''' Group Number (to dial group)''' 
| The number dialed to access this ring group. Must be 3 or 4 digits.
|-
| style="text-align: center;" | '''Allow Paging with **+Group Number''' 
| If enabled (checked), this will allow <nowiki>**+ </nowiki>group number to page this ring group
|-
| style="text-align: center;" | '''Ring Strategy''' 
| This defines how calls are distributed to member extensions. Options are:
'''Ring All''' - Ring all extensions in the group simultaneously

'''Round Robin''' - Ring extensions consecutively. Delivers a new call to the first person in the group only after the last person in the group has taken a call. If an extension is busy the call will automatically be routed to the next extension in the group.

'''Round Robin (with memory)''' - Ring extensions consecutively. Delivers a new call to the first person in the group only after the last person in the group has taken a call. Remembers where the last call was taken and distributes new calls to the next extension in the rotation.

'''Least Recent''' - Ring extensions in order of which was least recently called by the queue.

'''Fewest Calls''' - Ring extension in order of which has the fewest completed calls from the queue

'''Random''' - Distributes calls randomly to the group.

|-
| style="text-align: center;" | '''Failover Destination''' 
| This defines where a call will be sent when it goes unanswered for a duration that matches the Timeout for the group.
|-
| style="text-align: center;" | '''Timeout''' 
| Defines, in seconds, the total time a call will spend in the ring group before being sent to the Failover Destination
|-
| style="text-align: center;" | '''Allow Recording''' 
| Determines if feature code recording is allowed for calls in this queue.
If YES = enables the recording feature.

If NO = disables the recording feature.

|-
| style="text-align: center;" | '''Agent Ring Time''' 
| Defines, in seconds, how long the queue will ring a particular agent. If using the Ring All strategy, this value should match the Timeout. For other strategies, this should be set to how long an extension should ring before moving on to the next.
|-
| style="text-align: center;" | '''Autofill''' 
|
This deals with how the queue handles multiple concurrent calls. If set to NO the queue will wait until the first call is answered before ringing phones with the second call. If set to YES the queue will ring phones for all calls as they come in.  Note this setting has a different effect with ring all that may not be desirable. 

When autofill is yes and ring strategy is 'ringall' the group will still deliver the calls in the order they are received and in order to prevent calls from being lost by phones that have too few lines for the number of calls in the queue, the group will only deliver a single additional call when Ring In Use is set to yes. If someone who's phone is ringing rejects the call, the group will deliver the next call in line to all members.  This means the the rejected call is no longer ringing the member's phone that rejected the call, however it is still ringing all of the other members.  At the same time the 2nd call in line comes in and rings everyone.

|-
| style="text-align: center;" | '''Ring In use''' 
| If '''YES''' = Distribute calls to group members if they are already on a call
If '''NO''' = Do NOT distribute calls to group members if they are already on a call

|}

The following outlines the steps to add/edit ring group settings.

'''STEPS:'''

#From the '''Destinations'''=>'''Group''' page, click the '''ADD Group''' button. The '''Edit Ring Group''' page appears.
#In the '''Name''' field, enter a name for the group.
#In the '''Group Number''' field, enter a number for the group. This number must be three or four digits in length.
#Check the '''Allow Paging with <nowiki>**+</nowiki> the Group Number''' to enable the setting.
#Select the desired '''Ring Strategy''' from the drop-down list.
#Select the desired '''Failover Destination''' from the list of available options.
#Enter the time for the '''Timeout''' parameter.
#Select “'''Yes'''” if you want to enable the '''Allow Recording '''setting. Otherwise, set it to “'''No'''”.
#Enter the time for the '''Agent Ring Time''' parameter.
#Select “'''Yes'''” if you want to enable the '''Autofill '''setting. Otherwise, set it to “'''No'''”
#Select “'''Yes'''” if you want to enable the '''Ring in use '''setting. Otherwise, set it to “'''No'''”
#Click [[File:Savechanges.png]] the button.
#From the '''Members''' section of the '''Edit Ring Group''' page, select the extension number from the list of extensions that you want to add to the ring group. Click the '''ADD''' button.
#A second list appears next to the Members list displaying the selected extension. Repeat step 13 until all extensions that you want to add are on the list. Additionally, Shift or Control click functionality works for adding multiple extensions at once.
#To remove an item from the list of additions, select the extension number then click the '''REMOVE''' button.
#Define '''Priorities''' for the extensions, if needed. The scale ranges 0 thru 9 (0 is the highest priority).
#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.

==== Ring Group Advanced Settings ====

[[File:Customcallerid.png|center|Customcallerid.png]] The following table describes the parameters (fields) on the '''Advanced''' '''Settings''' for Ring Groups page and their descriptions and recommended settings.

{| class="wikitable"
|-
| colspan="2" | <center>'''Sections/Fields'''</center>
| <center>'''Description'''</center>
|-
| colspan="3" | <center>'''Ring/Hold Settings'''</center>
|-
| style="text-align: center;" | '''Caller Hears'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Override Hold Music'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Hold Music'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Ring Tone'''
| colspan="2" | When Caller Ring Settings are set to Ringing, use this to define the ring cadence.
|-
| colspan="3" | <center>'''Custom Caller ID Settings'''</center>
|-
| style="text-align: center;" | '''Use'''
| colspan="2" | Defines what method, if any, you wish to use to override Inbound CID. The options are:
System Default, Name, Number, Name and Number

|-
| style="text-align: center;" | '''Name'''
| colspan="2" | This is the name of the caller that will be displayed when a call is received.
|-
| style="text-align: center;" | '''Number'''
| colspan="2" | This is the number of the caller that will be displayed when a call is received.
|-
| style="text-align: center;" | '''Prepend'''
| colspan="2" | Selecting Prepend will enter the text you put in the field, followed by the original CID. If not prepending, the override eliminates the original information.
|}

==== Edit Advanced Ring Group Settings ====

'''STEPS:'''

#Select '''Destinations=>Groups'''. The '''Ring Groups''' page appears.
#Select the Ring Group you want to edit by either select the [[File:Penciledit.png]] icon to the right of the group name or click on the blue Group Name link.
#The '''Edit Ring Group''' page appears. Scroll down the page and click on the '''Advanced''' link. The '''Advanced Ring Group Settings''' section is displayed.
#Make the necessary changes to the settings then click button [[File:Savechanges.png]] 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.

==== Edit Custom Caller ID Settings ====

'''STEPS:'''

#Select '''Destinations=>Groups'''. The '''Ring Groups''' page appears.
#Select the Ring Group you want to edit by either select the [[File:Penciledit.png]] icon to the right of the group name or click on the blue Group Name link.
#The '''Edit Ring Group''' page appears. Scroll down the page and click on the '''Advanced''' link. '''The Custom Caller ID''' section is displayed.
#Make the necessary changes to the settings then click [[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.

==== Edit Members/Agents ====

[[File:Editmembersagents.png|center|Editmembersagents.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Members
| This is the list of available extensions that are in the database.
|-
| Agents
| This is the list of available agents that are in the database.
|-
| Priority
| This field allows you to set the priority level for the members or agents. The lower the number, the higher the priority.
|-
| ADD
| This button will add the select item on the list to the right of the available members or agents. Shift and Control click functionality works for adding more than one at a time.
|-
| REMOVE
| This button will remove the item(s) that have been selected from the list of available members or agents. Shift and Control click functionality works for adding more than one at a time.
|}

==== Add Agents/Members to the Group ====

'''STEPS:'''

#Navigate to the '''Destinations=>Groups=>Edit Group''' page.
#Select from the left hand column which agents/members to add to the group.
#Click the '''Add '''button.
#Click [[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 Agents/Members from the Group ====

'''STEPS:'''

#STEPS: Navigate to the '''Destinations'''=>'''Groups'''=>'''Edit Group '''page.
#Select from the right hand column which agents/members to remove from the group.
#Click the '''Add '''button.
#Click [[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

=== Automatic Call Distribution ([[ACD]]) ===

Systems licensed to use [[ACD]] will have access to additional functionality pertaining to Ring Groups. ACD allows for more flexibility and control over the queuing in a Ring Group, as well as allowing the creation of Agents who can be added to the Ring Group.

[[File:Autocalldistpage.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Agent Retry Timer
| This defines, in seconds, how long the queue will wait before attempting to ring an agent again.
|-
| Weight
| The priority of the queue in relation to other queues in the system.
|-
| Wrap-up Time
| This defines, in seconds, how long the queue will wait after an agent has ended a call before flagging then as an active agent ready to accept more calls.
|-
| Autopause
| This will place an agent in a pause state in relation to this queue when they ignore or answer a call. To become an active member of the queue again unpause using the 1* feature code a phone or 1*+agent# for an agent.
|-
| Maximum # of people in queue
| Defines how many callers can join the queue before it is considered full.
|-
| Announce Frequency
| Must be set for Announce Holdtime to function. This pertains to how often the caller will hear the hold time announcement, in seconds
|-
| Periodic Announce Frequency
| This defines how much time passes between playing the recording that was set for this queue under Periodic Announcement.
|-
| Announce Holdtime
| If set to YES this will announce a caller’s status in the queue with regards to their expected hold time, pending the hold time is estimated to be over 1 minute. If the hold time is less than one minute, this will announce the caller’s position in the queue. A period of time is needed for the algorithm to ‘learn’ and estimate hold times accurately. This message will not play for the first caller in the queue. Caller Ring Settings must be set to a MOH file in order for this message to play.
|-
| Join Empty Queue
| Controls if a caller can join a queue.
YES = Callers can join the queue with no members or whose members are busy.

NO = Callers can only join the queue with active, non-busy members.

STRICT = Callers can join the queue as long as there are active members, be they available or busy.

|-
| Leave Empty Queue
| Controls if a caller will leave an empty queue.
YES = Callers will leave the queue if there are no active members, or all members are busy.

NO = Callers will not leave the queue even if there are no active members, or all members are busy.

STRICT = Callers will leave the queue if there are no active members, but will remain in the queue if all members are busy.

|-
| Report Hold Time
| Set to YES if you want the person answering the queue call to hear a message indicating how long the call they just answered was in the queue
|-
| Member Delay before connect
| Delay, in seconds, before a member is connected to the queue.
|-
| Timeout Restart
| If set to YES the ring group timeout will reset after attempting to ring an agent/member. When set to NO, time taken to ring an agent/member will be subtracted from the ring group time, and when the timer ends the group will send the caller to the Failover Destination
|-
| Service Level
| Set this to the time, in seconds, that defines the acceptable level of service for the queue. The Live Queue Data page will track the percentage of calls answered within the allotted time.
|-
| Exit Status: FULL
Exit Status: JOINEMPTY

Exit Status: LEAVEEMPTY

Exit Status: JOINUNAVAIL

Exit Status: LEAVEUNAVAIL

| These are the destinations where calls will go based on the exit strategy defined. If no one in the group accepts the call this is the destination to which the call will be sent. Destinations can be extensions, other ring groups, a menu or a voicemail box.
|-
| Intro Announcement
| Define what prompt to play to the caller just before they enter the queue. (Previously “Announce Filename”)
|-
| Agent Announcement
| Defines what prompt to play to the agent/member who answers the queue call. This setting is set per Queue and therefore can be used to identify what Queue is being answered (for agents handling calls from multiple Queues.
|-
| Periodic Announcement
| Defines what prompt to play to the caller at the Periodic Announce frequency interval. This will not play for the caller in the first position of the queue. Caller Ring Settings must be set to a MOH file in order for this message to play.
|-
| Exit Menu
| Set this to a menu. Key presses while in this group will send people the corresponding destination from that menu.
|}

''Table 39Edit Ring Group ACD Agents Settings and Descriptions''

==== Edit Automatic Call Distribution (ACD) Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Make the necessary updates to the ACD settings.
#Click [[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.

==== Add Automatic Call Distribution (ACD) Agents ====

[[File:Addeditagentfeatures.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Name
| Enter a name that will identify this agent.
|-
| ID
| Enter a unique 3-4 digit number to identify this agent. This number is used to log in/out.
|-
| PIN
| Enter a minimum of 4 digits for the PIN. This is the password used to log in/out with the agent.
|}

''Table 40Add/Edit Agent Features and Descriptions''

 '''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The Agents page appears.
#Click on the '''ADD''' button to add a new agent. A box with the name, ID and PIN field appears. Enter the information for the new agent.
#Click [[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.

==== Edit Automatic Call Distribution (ACD) Agent ====

'''STEPS:'''

#From the '''Destinations''' '''Groups Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The '''Agents''' page appears.
#Update the necessary information for agent(s) then click the '''SAVE CHANGES''' button. The changes that you made will appear in the updated list.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

==== Delete Automatic Call Distribution (ACD) Agent ====

'''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The '''Agents''' page appears.
#From the list of existing agents, click on the box (placing a checkmark) to the left of the Agent(s) that you want to delete. Click the '''DELETE SELECTED''' button.
#The agent selected is removed from the list.
#Click [[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.

=== Multicast Paging Group ===

A Multicast Paging Group provides a way to page a large number of phones, while keeping the burden on the network and PBX to a minimum.

'''NOTE: IPitomy recommends using multicast paging for groups of 10 or more phones that are paged frequently.''' [[File:Multicastgroups.png|center|Multicastgroups.png]] [[File:Editmultipagewindow.png|center|Editmultipagewindow.png]] The following table describes the fields (parameters) on the Multicast Edit Paging Group window and the recommended settings.

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Name
| This is a name of the multicast paging group.
|-
| Local Number
| This is the number that will be dialed to launch the multicast paging to the defined group. This is the unique 3-4 digit number.
|-
| Multicast Address
| This should be a valid multicast IP for a Multicast Address<ref name="ftn1">224.0.0.0 to 239.255.255.255, with many addresses reserved.</ref>. We recommend '''225.5.5.5'''. You can use the same IP for each group, so long as you assign them different ports.
|-
| Multicast Port
| This must be a valid, unused port Multicast Port.
'''NOTE: This must be a port below 6535.'''

|-
| Time to Live
| Time to Live is a counter that decreases by 1 each time a packet passes through a router. When the counter reaches 0, the packet is considered dead and will not pass to any other networks. This can be left blank or configured as needed for your particular install.
'''NOTE: Leaving it blank the packet will not pass to another network.'''

|-
| Members
| The Members section allows you to add the extensions to be paged within the group when a page is sent.
|-
| Add/Remove
| The Add and Remove button allows you to add or remove members (extensions) from the paging group.
|-
| Create/Update
| Depending on the process that is being performed, this button will appear (be labeled) as either Create or Update. This will create a new member or process the changes made.
|-
| Close Window
| This link will close the Edit Page Group window and return you to the Ring Groups page.
|}

The following section outlines the steps to add a new Multicast Paging Group.

==== Add Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate to the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on the '''Add Groups '''button. The '''Edit Page Group''' page appears.
#Enter the necessary information for the new paging group.
#From the '''Members''' section of the '''Edit Page Group''' window, select the members (extensions) that you want to include in the paging group.

Use the CTRL or SHIFT button to select multiple or a range of extensions from the list.

#Click on the '''ADD''' button to add the selected members to the group.
#Click the '''CREATE''' button to save the changes. The “Group Created” message appears. Close the window (click on the “X”). The system returns you to the '''Ring Groups''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones that correspond to extensions that were added to the '''Multicast Paging Group'''. Once all the phones have been re-booted, you will be able to utilize this feature by simply dialing the Local Number that was just created.

==== Edit Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on [[File:Penciledit.png]] icon to the right of the '''Name '''of the paging group you want to update. The '''Edit Page Group''' page appears.
#Enter the necessary information for the paging group.
#Click the '''UPDATE''' button to save the changes. Close the window (click on the “X”). The system returns you to the '''Ring Groups''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones for all the extensions that were involved in the Multicast Paging Group.

==== Delete Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on [[File:Deleteselected.png]] icon to the right of the '''Name '''of the paging group you want to delete. The group is removed from the '''Multicast Paging Group''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones for all the extensions that were involved in the Multicast Paging Group which was deleted.

=== Live Queue Data ===

The Live Queue Data page displays current call statistics for ACD queues. The page will refresh every 10 seconds, allowing a manager to monitor the activity in the queue as it happens. You can Apply Changes if you would like to reset the data.

[[File:Ringlivequedata.png|center|Ringlivequedata.png]]

{| style="border-spacing: 0px;" class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| <queue selection box>
| In the queue selection box the Queue or Queues to be displayed can be selected. Use the Control key (Command on Mac) to select multiple queues for display.
|-
| <queue statistics box>
| The Queue Statistics box has been reformatted in SW release 3.4.1 to allow its use on call center wall boards (large PC displays).
Each queue is encapsulated with a thin-line box. Queue statistics are color-coded for quick orientation on wall boards. Since this data is presented in GUI format it can be easily resized to accommodate large displays.

The top line of the queue display presents the Queue Name and the date.

Statistics are a running total and will reset to zero whenever Apply Changes is clicked (as is the case of the screen capture above).

|-
| Time in Q
| This is the average time that callers have waited in queue before being answered by a queue member/agent.
|-
| Calls in Q
| This is the number of calls that are waiting in the queue.
|-
| Answered
| This is the number of calls answered by members/agent of this queue.
|-
| Abandoned
| This is the number of calls that have not been answered and the caller abandoned the queue.
|-
| Target Time
| Displays the service level set for this ACD queue. The Service Level is the arbitrary time allocated by the queue administrator which is set as a gauge by which all calls should be answered.
|-
|  % in Target
| This is the percentage of answered calls measured against the service level. (The percentage of calls being answered within the set Target Time.)
|-
| <Member/Agent field>
| At the right side of the Queue Statistics box the agents and queue members are listed. There are two maximum columns and the window will resize to accommodate all members and agents registered to this queue.
Member/Agent status is available at a glance with color-codes as follows:

{| style="border-spacing:0;"
|-
| style="background-color:#c00000;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Red'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | In Use
|-
| style="background-color:#006600;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Green'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Idle (Logged In)
|-
| style="background-color:#ffff00;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Yellow'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Ringing
|-
| style="background-color:#7f7f7f;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Gray'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Unavailable (Not Logged In)
|-
| style="background-color:#e36c0a;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Orange'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Paused
|}

Also in the Member/Agent data are two numeral fields. The first number represents the Priority of this Member/Agent in this queue. The second number is the total calls taken by this Member/Agent.

|}

=== Live Queue Data Wallboard ===

Software version 3.4.1 and above allows the use of call center wall boards with Live Call Queue. This function is facilitated using a special user name and password via Smart Personal Console.

==== Activate Live Call Queue Wallboard ====

'''STEPS:'''

#Navigate to the '''PBX Setup'''=>'''General '''page
#Locate to the '''General Settings''' section (second section from the top).
#Locate the SPC Reports User Password field (bottom of section)
[[File:Spcreportsuserpass.png]]

#It may be helpful to change this to a user-friendly password. If you change the password be sure to [[File:Savechanges.png]] and '''Apply Changes'''.
#Make a note of this password.
#On the PC that will be used for the wallboard display, open a web browser and navigate to the IPitomy PBX login screen
#Using the right sideSmart Personal Console (User Login) input the word “reports” as the user name and the password saved in step 3. [[File:Reportslogin.png]]
#Click on '''Login''' .
#The CDR-Report page is opened. On this page is a button to select Live Queue.
#Click on '''Live Queue'''.
#The same display presented in PBX Administration is displayed. Simply select the group or groups that you wish to display on the wallboard.
#Size this web page data output for the monitor in use by pressing and holding the Cntrl key (PC, Command key on Mac) and press “+” or “-“ until the data is sized as you wish.

----

<references/>

IP PBX Manual Destinations Groups

2013-06-25T12:31:28Z

Nick Branica:

{{IP_PBX_Manual|sortkey=Groups}}
'''Groups'''

A Group is a logical grouping of extensions and their privileges. Each Group is associated with a set of PBX features and call routing strategies. It is advised to build the extensions in the PBX first, allowing the groups to be populated at the time of creation.

These Groups allow incoming calls to be distributed to a group of extensions rather than just one extension. Within the Groups, different distribution strategies may be selected based on the call coverage required by the application. Groups also provide off-hook paging functionality. By dialing the intercom button or code followed by the group number, extensions in the group will receive the page over the intercom. Groups are a set of extensions that are related either because they:

* Serve a similar business function.
* Work within the same department.
* Are located in proximity to each other.

For example, a business might create a few user Groups in an IP telephony network, e.g. Sales, Marketing, Administration, Accounting, Engineering, etc. Groups may also be created for people in similar locations like a plant floor, the north section of a building or the front office. The goal of a group is to ring telephones based on the incoming routing logic (e.g. DIDs, Auto Attendant selected by the incoming caller, or time of day). Note that calls routing to an extension from a ring group will ignore that phones PBX forwarding settings. Some phones have forwarding settings which are independent of the PBX (not set through the PBX Web GUI). These settings will still be applied.

* Link to ''Group'' training video - [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

[[File:ringgroupspages.png|center]]

=== Ring Group Examples ===
Ring groups define a set of extensions (people) that answer calls. Ring groups can be created for departments (e.g. Sales, Engineering) or buisiness regions (e.g. north, south, etc), or areas of a business (e.g. a warehouse or plant floor). These ring groups can appear on an Automated Attendant (menu). When the group is selected from a Menu, the call is routed to the group. These calls will be distributed to the member extensions based upon the ring strategy for the group. The ring strategy for the group can be set from the drop-down list. Available call distribution options are:

* Ring AllRing all extensions in the group simultaneously
* Round RobinTake turns ringing each extension, starts at the first extension for each new call
* Round Robin (with memory)Take turns ringing each extension, starts where the last call left off
* Least RecentRing extensions in order of which was least recently called by the queue. If using agents, logging out will reset the call counter so the next time that agent logs in, they will be at the top of the list for queue calls.
* Fewest CallsRing extension in order of which has the fewest completed calls from the queue
* RandomRandomly ring one extension at a time

==== Example Ring Group 1Departmental Grouping ====
Ring group definitions can be found in the '''Add a New Ring Group''' section of this guide. If a business has the following departments and people: This business can setup the following ring groups supporting business operations.

{| style="border-spacing:0;"
| style="background-color:#4f81bd;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;"| '''GROUP 1 - Sales'''
| style="background-color:#4f81bd;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;"| '''Ext.'''
| style="background-color:#c0504d;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;"| '''GROUP 2Customer Services'''
| style="background-color:#c0504d;border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| '''Ext.'''

|-
| 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;"| Cathy Caldwell
| 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;"| 123
| 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;"| Gretchen Goodall
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| 134

|-
| 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;"| David Dawson
| 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;"| 124
| 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;"| Peter Polk
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| 135

|-
| 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;"| Susan Smith
| 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;"| 125
| 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;"|

|-
| 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;"| Robert Reed
| 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;"| 126
| 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;"|

|}
Using this ring group scenario the '''Menu would look like''':

* Sales (Destination - Group 1).
* Customer Service (Destination - Group 2).
* Office Manager (Destination - Ext. 113).

The menu prompt (message) for this menu and group arrangement would read as follows:

'''“Thank you for calling CompanyA, a leading manufacturer of cable assemblies and wiring harnesses. If you know the extension of the party you would like to reach you may dial it at any time.”'''

* For Sales, press 1.
* For Customer Service, press 2.
* For Accounting, press 3.

Once a call is sent to Sales, member extension will be rang according to the '''ring group strategy.'''

==== Example Ring Group 2Regional Sales Grouping ====

{| style="border-spacing:0;"
| style="background-color:#4f81bd;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;"| '''GROUP 1 East Coast Sales'''
| style="background-color:#4f81bd;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;"| '''Ext.'''
| style="background-color:#c0504d;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;"| '''GROUP 2 -West Coast Sales'''
| style="background-color:#c0504d;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;"| '''Ext.'''
| style="background-color:#9bbb59;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;"| '''GROUP 3 Customer Service'''
| style="background-color:#9bbb59;border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| '''Ext.'''

|-
| 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;"| Cathy Caldwell
| 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;"| 123
| 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;"| Susan Smith
| 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;"| 125
| 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;"| Gretchen Goodall
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| 134

|-
| 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;"| David Dawson
| 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;"| 124
| 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;"| Robert Reed
| 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;"| 126
| 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;"| Peter Polk
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| 135

|}
Using this ring group scenario the '''Menu would look like''':

* East Coast Sales (Destination - Group 1).
* West Coast Sales (Destination - Group 2).
* Customer Service (Destination - Group 3).
* Office Manager (Destination - Ext. 113).

The menu '''prompt '''for this menu and group arrangement would read as follows:

'''“Thank you for calling CompanyA, a leading manufacturer of cable assemblies and wiring harnesses. If you know the extension of the party you would like to reach you may dial it at any time. For”'''

* East Coast Sales, press 1.
* West Coast Sales, press 2.
* Customer Service, press 3.
* Accounting, press 4.

Calls sent to these groups might use different ring strategies. The East Coast Sales group might answer calls Round Robin, distributing the calls to each Sale Representative consecutively. If a sales person is missing from the West Coast team this group might set phones to Ring All in the group so that calls don’t get missed. The Customer Service team may get high volumes of calls during a specified period of time, so this group may be set to ring Least Recent.

==== Add/Edit New Ring Group ====
[[File:editringgroup.png|center]]
The following table describes the parameters (fields) on the Edit Ring Group page and their descriptions and recommended settings.

{| border="1" class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| style="text-align: center;" | '''Name''' 
| The name associated with this ring group.
|-
| style="text-align: center;" | ''' Group Number (to dial group)''' 
| The number dialed to access this ring group. Must be 3 or 4 digits.
|-
| style="text-align: center;" | '''Allow Paging with **+Group Number''' 
| If enabled (checked), this will allow <nowiki>**+ </nowiki>group number to page this ring group
|-
| style="text-align: center;" | '''Ring Strategy''' 
| This defines how calls are distributed to member extensions. Options are:
'''Ring All''' - Ring all extensions in the group simultaneously

'''Round Robin''' - Ring extensions consecutively. Delivers a new call to the first person in the group only after the last person in the group has taken a call. If an extension is busy the call will automatically be routed to the next extension in the group.

'''Round Robin (with memory)''' - Ring extensions consecutively. Delivers a new call to the first person in the group only after the last person in the group has taken a call. Remembers where the last call was taken and distributes new calls to the next extension in the rotation.

'''Least Recent''' - Ring extensions in order of which was least recently called by the queue.

'''Fewest Calls''' - Ring extension in order of which has the fewest completed calls from the queue

'''Random''' - Distributes calls randomly to the group.

|-
| style="text-align: center;" | '''Failover Destination''' 
| This defines where a call will be sent when it goes unanswered for a duration that matches the Timeout for the group.
|-
| style="text-align: center;" | '''Timeout''' 
| Defines, in seconds, the total time a call will spend in the ring group before being sent to the Failover Destination
|-
| style="text-align: center;" | '''Allow Recording''' 
| Determines if feature code recording is allowed for calls in this queue.
If YES = enables the recording feature.

If NO = disables the recording feature.

|-
| style="text-align: center;" | '''Agent Ring Time''' 
| Defines, in seconds, how long the queue will ring a particular agent. If using the Ring All strategy, this value should match the Timeout. For other strategies, this should be set to how long an extension should ring before moving on to the next.
|-
| style="text-align: center;" | '''Autofill''' 
|
This deals with how the queue handles multiple concurrent calls. If set to NO the queue will wait until the first call is answered before ringing phones with the second call. If set to YES the queue will ring phones for all calls as they come in.  Note this setting has a different effect with ring all that may not be desirable. 

When autofill is yes and ring strategy is 'ringall' the group will still deliver the calls in the order they are received and in order to prevent calls from being lost by phones that have too few lines for the number of calls in the queue, the group will only deliver a single additional call when Ring In Use is set to yes. If someone who's phone is ringing rejects the call, the group will deliver the next call in line to all members.  This means the the rejected call is no longer ringing the member's phone that rejected the call, however it is still ringing all of the other members.  At the same time the 2nd call in line comes in and rings everyone.

|-
| style="text-align: center;" | '''Ring In use''' 
| If '''YES''' = Distribute calls to group members if they are already on a call
If '''NO''' = Do NOT distribute calls to group members if they are already on a call

|}

The following outlines the steps to add/edit ring group settings.

'''STEPS:'''

#From the '''Destinations'''=>'''Group''' page, click the '''ADD Group''' button. The '''Edit Ring Group''' page appears.
#In the '''Name''' field, enter a name for the group.
#In the '''Group Number''' field, enter a number for the group. This number must be three or four digits in length.
#Check the '''Allow Paging with <nowiki>**+</nowiki> the Group Number''' to enable the setting.
#Select the desired '''Ring Strategy''' from the drop-down list.
#Select the desired '''Failover Destination''' from the list of available options.
#Enter the time for the '''Timeout''' parameter.
#Select “'''Yes'''” if you want to enable the '''Allow Recording '''setting. Otherwise, set it to “'''No'''”.
#Enter the time for the '''Agent Ring Time''' parameter.
#Select “'''Yes'''” if you want to enable the '''Autofill '''setting. Otherwise, set it to “'''No'''”
#Select “'''Yes'''” if you want to enable the '''Ring in use '''setting. Otherwise, set it to “'''No'''”
#Click [[File:savechanges.png]] the button.
#From the '''Members''' section of the '''Edit Ring Group''' page, select the extension number from the list of extensions that you want to add to the ring group. Click the '''ADD''' button.
#A second list appears next to the Members list displaying the selected extension. Repeat step 13 until all extensions that you want to add are on the list. Additionally, Shift or Control click functionality works for adding multiple extensions at once.
#To remove an item from the list of additions, select the extension number then click the '''REMOVE''' button.
#Define '''Priorities''' for the extensions, if needed. The scale ranges 0 thru 9 (0 is the highest priority).
#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.

==== Ring Group Advanced Settings ====
[[File:customcallerid.png|center]]
The following table describes the parameters (fields) on the '''Advanced''' '''Settings''' for Ring Groups page and their descriptions and recommended settings.

{| class="wikitable"
|-
| colspan="2" | <center>'''Sections/Fields'''</center>
| <center>'''Description'''</center>
|-
| colspan="3" | <center>'''Ring/Hold Settings'''</center>
|-
| style="text-align: center;" | '''Caller Hears'''
| colspan="2" | update text

|-
| style="text-align: center;" | '''Override Hold Music'''
| colspan="2" | update text
|-
|style="text-align: center;" | '''Hold Music'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Ring Tone'''
| colspan="2" | When Caller Ring Settings are set to Ringing, use this to define the ring cadence.
|-
| colspan="3" | <center>'''Custom Caller ID Settings'''</center>
|-
| style="text-align: center;" | '''Use'''
| colspan="2" | Defines what method, if any, you wish to use to override Inbound CID. The options are:
System Default, Name, Number, Name and Number

|-
| style="text-align: center;" | '''Name'''
| colspan="2" | This is the name of the caller that will be displayed when a call is received.
|-
| style="text-align: center;" | '''Number'''
| colspan="2" | This is the number of the caller that will be displayed when a call is received.
|-
| style="text-align: center;" | '''Prepend'''
| colspan="2" | Selecting Prepend will enter the text you put in the field, followed by the original CID. If not prepending, the override eliminates the original information.
|}

==== Edit Advanced Ring Group Settings ====
'''STEPS:'''

# Select '''Destinations=>Groups'''. The '''Ring Groups''' page appears.
# Select the Ring Group you want to edit by either select the [[File:penciledit.png]] icon to the right of the group name or click on the blue Group Name link.
# The '''Edit Ring Group''' page appears. Scroll down the page and click on the '''Advanced''' link. The '''Advanced Ring Group Settings''' section is displayed.
# Make the necessary changes to the settings then click button [[File:savechanges.png]] 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.

==== Edit Custom Caller ID Settings ====
'''STEPS:'''

# Select '''Destinations=>Groups'''. The '''Ring Groups''' page appears.
# Select the Ring Group you want to edit by either select the [[File:penciledit.png]] icon to the right of the group name or click on the blue Group Name link.
# The '''Edit Ring Group''' page appears. Scroll down the page and click on the '''Advanced''' link. '''The Custom Caller ID''' section is displayed.
# Make the necessary changes to the settings then click [[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.

==== Edit Members/Agents ====
[[File:editmembersagents.png|center]]
{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Members
| This is the list of available extensions that are in the database.
|-
| Agents
| This is the list of available agents that are in the database.
|-
| Priority
| This field allows you to set the priority level for the members or agents. The lower the number, the higher the priority.
|-
| ADD
| This button will add the select item on the list to the right of the available members or agents. Shift and Control click functionality works for adding more than one at a time.
|-
| REMOVE
| This button will remove the item(s) that have been selected from the list of available members or agents. Shift and Control click functionality works for adding more than one at a time.
|}

==== Add Agents/Members to the Group ====
'''STEPS:'''

# Navigate to the '''Destinations=>Groups=>Edit Group''' page.
# Select from the left hand column which agents/members to add to the group.
# Click the '''Add '''button.
# Click [[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 Agents/Members from the Group ====
'''STEPS:'''

# STEPS: Navigate to the '''Destinations'''=>'''Groups'''=>'''Edit Group '''page.
# Select from the right hand column which agents/members to remove from the group.
# Click the '''Add '''button.
# Click [[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

=== Automatic Call Distribution ([[ACD]]) ===

Systems licensed to use [[ACD]] will have access to additional functionality pertaining to Ring Groups. ACD allows for more flexibility and control over the queuing in a Ring Group, as well as allowing the creation of Agents who can be added to the Ring Group.

[[File:autocalldistpage.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Agent Retry Timer
| This defines, in seconds, how long the queue will wait before attempting to ring an agent again.
|-
| Weight
| The priority of the queue in relation to other queues in the system.
|-
| Wrap-up Time
| This defines, in seconds, how long the queue will wait after an agent has ended a call before flagging then as an active agent ready to accept more calls.
|-
| Autopause
| This will place an agent in a pause state in relation to this queue when they ignore or answer a call. To become an active member of the queue again unpause using the 1* feature code a phone or 1*+agent# for an agent.
|-
| Maximum # of people in queue
| Defines how many callers can join the queue before it is considered full.
|-
| Announce Frequency
| Must be set for Announce Holdtime to function. This pertains to how often the caller will hear the hold time announcement, in seconds
|-
| Periodic Announce Frequency
| This defines how much time passes between playing the recording that was set for this queue under Periodic Announcement.
|-
| Announce Holdtime
| If set to YES this will announce a caller’s status in the queue with regards to their expected hold time, pending the hold time is estimated to be over 1 minute. If the hold time is less than one minute, this will announce the caller’s position in the queue. A period of time is needed for the algorithm to ‘learn’ and estimate hold times accurately. This message will not play for the first caller in the queue. Caller Ring Settings must be set to a MOH file in order for this message to play.
|-
| Join Empty Queue
| Controls if a caller can join a queue.
YES = Callers can join the queue with no members or whose members are busy.

NO = Callers can only join the queue with active, non-busy members.

STRICT = Callers can join the queue as long as there are active members, be they available or busy.

|-
| Leave Empty Queue
| Controls if a caller will leave an empty queue.
YES = Callers will leave the queue if there are no active members, or all members are busy.

NO = Callers will not leave the queue even if there are no active members, or all members are busy.

STRICT = Callers will leave the queue if there are no active members, but will remain in the queue if all members are busy.

|-
| Report Hold Time
| Set to YES if you want the person answering the queue call to hear a message indicating how long the call they just answered was in the queue
|-
| Member Delay before connect
| Delay, in seconds, before a member is connected to the queue.
|-
| Timeout Restart
| If set to YES the ring group timeout will reset after attempting to ring an agent/member. When set to NO, time taken to ring an agent/member will be subtracted from the ring group time, and when the timer ends the group will send the caller to the Failover Destination
|-
| Service Level
| Set this to the time, in seconds, that defines the acceptable level of service for the queue. The Live Queue Data page will track the percentage of calls answered within the allotted time.
|-
| Exit Status: FULL
Exit Status: JOINEMPTY

Exit Status: LEAVEEMPTY

Exit Status: JOINUNAVAIL

Exit Status: LEAVEUNAVAIL

| These are the destinations where calls will go based on the exit strategy defined. If no one in the group accepts the call this is the destination to which the call will be sent. Destinations can be extensions, other ring groups, a menu or a voicemail box.
|-
| Intro Announcement
| Define what prompt to play to the caller just before they enter the queue. (Previously “Announce Filename”)
|-
| Agent Announcement
| Defines what prompt to play to the agent/member who answers the queue call. This setting is set per Queue and therefore can be used to identify what Queue is being answered (for agents handling calls from multiple Queues.
|-
| Periodic Announcement
| Defines what prompt to play to the caller at the Periodic Announce frequency interval. This will not play for the caller in the first position of the queue. Caller Ring Settings must be set to a MOH file in order for this message to play.
|-
| Exit Menu
| Set this to a menu. Key presses while in this group will send people the corresponding destination from that menu.
|}

''Table 39Edit Ring Group ACD Agents Settings and Descriptions''

==== Edit Automatic Call Distribution (ACD) Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Make the necessary updates to the ACD settings.
#Click [[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.

==== Add Automatic Call Distribution (ACD) Agents ====
[[File:addeditagentfeatures.png]]
{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Name
| Enter a name that will identify this agent.
|-
| ID
| Enter a unique 3-4 digit number to identify this agent. This number is used to log in/out.
|-
| PIN
| Enter a minimum of 4 digits for the PIN. This is the password used to log in/out with the agent.
|}

''Table 40Add/Edit Agent Features and Descriptions''

 '''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The Agents page appears.
#Click on the '''ADD''' button to add a new agent. A box with the name, ID and PIN field appears. Enter the information for the new agent.
#Click [[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.

==== Edit Automatic Call Distribution (ACD) Agent ====

'''STEPS:'''

#From the '''Destinations''' '''Groups Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The '''Agents''' page appears.
#Update the necessary information for agent(s) then click the '''SAVE CHANGES''' button. The changes that you made will appear in the updated list.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

==== Delete Automatic Call Distribution (ACD) Agent ====

'''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The '''Agents''' page appears.
#From the list of existing agents, click on the box (placing a checkmark) to the left of the Agent(s) that you want to delete. Click the '''DELETE SELECTED''' button.
#The agent selected is removed from the list.
#Click [[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.

=== Multicast Paging Group ===

A Multicast Paging Group provides a way to page a large number of phones, while keeping the burden on the network and PBX to a minimum.

'''NOTE: IPitomy recommends using multicast paging for groups of 10 or more phones that are paged frequently.'''
[[File:multicastgroups.png|center]]
[[File:editmultipagewindow.png|center]]
 The following table describes the fields (parameters) on the Multicast Edit Paging Group window and the recommended settings.

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Name
| This is a name of the multicast paging group.
|-
| Local Number
| This is the number that will be dialed to launch the multicast paging to the defined group. This is the unique 3-4 digit number.
|-
| Multicast Address
| This should be a valid multicast IP for a Multicast Address<ref name="ftn1">224.0.0.0 to 239.255.255.255, with many addresses reserved.</ref>. We recommend '''225.5.5.5'''. You can use the same IP for each group, so long as you assign them different ports.
|-
| Multicast Port
| This must be a valid, unused port Multicast Port.
'''NOTE: This must be a port below 6535.'''

|-
| Time to Live
| Time to Live is a counter that decreases by 1 each time a packet passes through a router. When the counter reaches 0, the packet is considered dead and will not pass to any other networks. This can be left blank or configured as needed for your particular install.
'''NOTE: Leaving it blank the packet will not pass to another network.'''

|-
| Members
| The Members section allows you to add the extensions to be paged within the group when a page is sent.
|-
| Add/Remove
| The Add and Remove button allows you to add or remove members (extensions) from the paging group.
|-
| Create/Update
| Depending on the process that is being performed, this button will appear (be labeled) as either Create or Update. This will create a new member or process the changes made.
|-
| Close Window
| This link will close the Edit Page Group window and return you to the Ring Groups page.
|}

The following section outlines the steps to add a new Multicast Paging Group.

==== Add Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate to the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on the '''Add Groups '''button. The '''Edit Page Group''' page appears.
#Enter the necessary information for the new paging group.
#From the '''Members''' section of the '''Edit Page Group''' window, select the members (extensions) that you want to include in the paging group.

Use the CTRL or SHIFT button to select multiple or a range of extensions from the list.

#Click on the '''ADD''' button to add the selected members to the group.
#Click the '''CREATE''' button to save the changes. The “Group Created” message appears. Close the window (click on the “X”). The system returns you to the '''Ring Groups''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones that correspond to extensions that were added to the '''Multicast Paging Group'''. Once all the phones have been re-booted, you will be able to utilize this feature by simply dialing the Local Number that was just created.

==== Edit Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on [[File:penciledit.png]] icon to the right of the '''Name '''of the paging group you want to update. The '''Edit Page Group''' page appears.
#Enter the necessary information for the paging group.
#Click the '''UPDATE''' button to save the changes. Close the window (click on the “X”). The system returns you to the '''Ring Groups''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones for all the extensions that were involved in the Multicast Paging Group.

==== Delete Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on [[File:deleteselected.png]] icon to the right of the '''Name '''of the paging group you want to delete. The group is removed from the '''Multicast Paging Group''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones for all the extensions that were involved in the Multicast Paging Group which was deleted.

=== Live Queue Data ===

The Live Queue Data page displays current call statistics for ACD queues. The page will refresh every 10 seconds, allowing a manager to monitor the activity in the queue as it happens. You can Apply Changes if you would like to reset the data.

[[File:ringlivequedata.png|center]]

{| style="border-spacing: 0px;" class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| <queue selection box>
| In the queue selection box the Queue or Queues to be displayed can be selected. Use the Control key (Command on Mac) to select multiple queues for display.
|-
| <queue statistics box>
| The Queue Statistics box has been reformatted in SW release 3.4.1 to allow its use on call center wall boards (large PC displays).
Each queue is encapsulated with a thin-line box. Queue statistics are color-coded for quick orientation on wall boards. Since this data is presented in GUI format it can be easily resized to accommodate large displays.

The top line of the queue display presents the Queue Name and the date.

Statistics are a running total and will reset to zero whenever Apply Changes is clicked (as is the case of the screen capture above).

|-
| Time in Q
| This is the average time that callers have waited in queue before being answered by a queue member/agent.
|-
| Calls in Q
| This is the number of calls that are waiting in the queue.
|-
| Answered
| This is the number of calls answered by members/agent of this queue.
|-
| Abandoned
| This is the number of calls that have not been answered and the caller abandoned the queue.
|-
| Target Time
| Displays the service level set for this ACD queue. The Service Level is the arbitrary time allocated by the queue administrator which is set as a gauge by which all calls should be answered.
|-
|  % in Target
| This is the percentage of answered calls measured against the service level. (The percentage of calls being answered within the set Target Time.)
|-
| <Member/Agent field>
| At the right side of the Queue Statistics box the agents and queue members are listed. There are two maximum columns and the window will resize to accommodate all members and agents registered to this queue.
Member/Agent status is available at a glance with color-codes as follows:

{| style="border-spacing:0;"
|-
| style="background-color:#c00000;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Red'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | In Use
|-
| style="background-color:#006600;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Green'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Idle (Logged In)
|-
| style="background-color:#ffff00;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Yellow'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Ringing
|-
| style="background-color:#7f7f7f;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Gray'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Unavailable (Not Logged In)
|-
| style="background-color:#e36c0a;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Orange'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Paused
|}

Also in the Member/Agent data are two numeral fields. The first number represents the Priority of this Member/Agent in this queue. The second number is the total calls taken by this Member/Agent.

|}

=== Live Queue Data Wallboard ===
Software version 3.4.1 and above allows the use of call center wall boards with Live Call Queue. This function is facilitated using a special user name and password via Smart Personal Console.

==== Activate Live Call Queue Wallboard ====
'''STEPS:'''

# Navigate to the '''PBX Setup'''=>'''General '''page
# Locate to the '''General Settings''' section (second section from the top).
# Locate the SPC Reports User Password field (bottom of section)[[File:spcreportsuserpass.png]]
# It may be helpful to change this to a user-friendly password. If you change the password be sure to [[File:savechanges.png]] and '''Apply Changes'''.
# Make a note of this password.
# On the PC that will be used for the wallboard display, open a web browser and navigate to the IPitomy PBX login screen
# Using the right sideSmart Personal Console (User Login) input the word “reports” as the user name and the password saved in step 3. [[File:reportslogin.png]]
# Click on '''Login''' .
# The CDR-Report page is opened. On this page is a button to select Live Queue.
# Click on '''Live Queue'''.
# The same display presented in PBX Administration is displayed. Simply select the group or groups that you wish to display on the wallboard.
# Size this web page data output for the monitor in use by pressing and holding the Cntrl key (PC, Command key on Mac) and press “+” or “-“ until the data is sized as you wish.

----
<references/>

IP PBX Manual Destinations Groups

2013-06-25T12:29:18Z

Nick Branica:

{{IP_PBX_Manual|sortkey=Groups}}
'''Groups'''

A Group is a logical grouping of extensions and their privileges. Each Group is associated with a set of PBX features and call routing strategies. It is advised to build the extensions in the PBX first, allowing the groups to be populated at the time of creation.

These Groups allow incoming calls to be distributed to a group of extensions rather than just one extension. Within the Groups, different distribution strategies may be selected based on the call coverage required by the application. Groups also provide off-hook paging functionality. By dialing the intercom button or code followed by the group number, extensions in the group will receive the page over the intercom. Groups are a set of extensions that are related either because they:

* Serve a similar business function.
* Work within the same department.
* Are located in proximity to each other.

For example, a business might create a few user Groups in an IP telephony network, e.g. Sales, Marketing, Administration, Accounting, Engineering, etc. Groups may also be created for people in similar locations like a plant floor, the north section of a building or the front office. The goal of a group is to ring telephones based on the incoming routing logic (e.g. DIDs, Auto Attendant selected by the incoming caller, or time of day). Note that calls routing to an extension from a ring group will ignore that phones PBX forwarding settings. Some phones have forwarding settings which are independent of the PBX (not set through the PBX Web GUI). These settings will still be applied.

* Link to Group training video - [https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

[[File:ringgroupspages.png|center]]

=== Ring Group Examples ===
Ring groups define a set of extensions (people) that answer calls. Ring groups can be created for departments (e.g. Sales, Engineering) or buisiness regions (e.g. north, south, etc), or areas of a business (e.g. a warehouse or plant floor). These ring groups can appear on an Automated Attendant (menu). When the group is selected from a Menu, the call is routed to the group. These calls will be distributed to the member extensions based upon the ring strategy for the group. The ring strategy for the group can be set from the drop-down list. Available call distribution options are:

* Ring AllRing all extensions in the group simultaneously
* Round RobinTake turns ringing each extension, starts at the first extension for each new call
* Round Robin (with memory)Take turns ringing each extension, starts where the last call left off
* Least RecentRing extensions in order of which was least recently called by the queue. If using agents, logging out will reset the call counter so the next time that agent logs in, they will be at the top of the list for queue calls.
* Fewest CallsRing extension in order of which has the fewest completed calls from the queue
* RandomRandomly ring one extension at a time

==== Example Ring Group 1Departmental Grouping ====
Ring group definitions can be found in the '''Add a New Ring Group''' section of this guide. If a business has the following departments and people: This business can setup the following ring groups supporting business operations.

{| style="border-spacing:0;"
| style="background-color:#4f81bd;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;"| '''GROUP 1 - Sales'''
| style="background-color:#4f81bd;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;"| '''Ext.'''
| style="background-color:#c0504d;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;"| '''GROUP 2Customer Services'''
| style="background-color:#c0504d;border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| '''Ext.'''

|-
| 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;"| Cathy Caldwell
| 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;"| 123
| 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;"| Gretchen Goodall
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| 134

|-
| 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;"| David Dawson
| 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;"| 124
| 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;"| Peter Polk
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| 135

|-
| 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;"| Susan Smith
| 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;"| 125
| 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;"|

|-
| 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;"| Robert Reed
| 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;"| 126
| 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;"|

|}
Using this ring group scenario the '''Menu would look like''':

* Sales (Destination - Group 1).
* Customer Service (Destination - Group 2).
* Office Manager (Destination - Ext. 113).

The menu prompt (message) for this menu and group arrangement would read as follows:

'''“Thank you for calling CompanyA, a leading manufacturer of cable assemblies and wiring harnesses. If you know the extension of the party you would like to reach you may dial it at any time.”'''

* For Sales, press 1.
* For Customer Service, press 2.
* For Accounting, press 3.

Once a call is sent to Sales, member extension will be rang according to the '''ring group strategy.'''

==== Example Ring Group 2Regional Sales Grouping ====

{| style="border-spacing:0;"
| style="background-color:#4f81bd;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;"| '''GROUP 1 East Coast Sales'''
| style="background-color:#4f81bd;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;"| '''Ext.'''
| style="background-color:#c0504d;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;"| '''GROUP 2 -West Coast Sales'''
| style="background-color:#c0504d;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;"| '''Ext.'''
| style="background-color:#9bbb59;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;"| '''GROUP 3 Customer Service'''
| style="background-color:#9bbb59;border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| '''Ext.'''

|-
| 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;"| Cathy Caldwell
| 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;"| 123
| 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;"| Susan Smith
| 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;"| 125
| 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;"| Gretchen Goodall
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| 134

|-
| 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;"| David Dawson
| 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;"| 124
| 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;"| Robert Reed
| 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;"| 126
| 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;"| Peter Polk
| style="border:0.0069in solid #000000;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;"| 135

|}
Using this ring group scenario the '''Menu would look like''':

* East Coast Sales (Destination - Group 1).
* West Coast Sales (Destination - Group 2).
* Customer Service (Destination - Group 3).
* Office Manager (Destination - Ext. 113).

The menu '''prompt '''for this menu and group arrangement would read as follows:

'''“Thank you for calling CompanyA, a leading manufacturer of cable assemblies and wiring harnesses. If you know the extension of the party you would like to reach you may dial it at any time. For”'''

* East Coast Sales, press 1.
* West Coast Sales, press 2.
* Customer Service, press 3.
* Accounting, press 4.

Calls sent to these groups might use different ring strategies. The East Coast Sales group might answer calls Round Robin, distributing the calls to each Sale Representative consecutively. If a sales person is missing from the West Coast team this group might set phones to Ring All in the group so that calls don’t get missed. The Customer Service team may get high volumes of calls during a specified period of time, so this group may be set to ring Least Recent.

==== Add/Edit New Ring Group ====
[[File:editringgroup.png|center]]
The following table describes the parameters (fields) on the Edit Ring Group page and their descriptions and recommended settings.

{| border="1" class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| style="text-align: center;" | '''Name''' 
| The name associated with this ring group.
|-
| style="text-align: center;" | ''' Group Number (to dial group)''' 
| The number dialed to access this ring group. Must be 3 or 4 digits.
|-
| style="text-align: center;" | '''Allow Paging with **+Group Number''' 
| If enabled (checked), this will allow <nowiki>**+ </nowiki>group number to page this ring group
|-
| style="text-align: center;" | '''Ring Strategy''' 
| This defines how calls are distributed to member extensions. Options are:
'''Ring All''' - Ring all extensions in the group simultaneously

'''Round Robin''' - Ring extensions consecutively. Delivers a new call to the first person in the group only after the last person in the group has taken a call. If an extension is busy the call will automatically be routed to the next extension in the group.

'''Round Robin (with memory)''' - Ring extensions consecutively. Delivers a new call to the first person in the group only after the last person in the group has taken a call. Remembers where the last call was taken and distributes new calls to the next extension in the rotation.

'''Least Recent''' - Ring extensions in order of which was least recently called by the queue.

'''Fewest Calls''' - Ring extension in order of which has the fewest completed calls from the queue

'''Random''' - Distributes calls randomly to the group.

|-
| style="text-align: center;" | '''Failover Destination''' 
| This defines where a call will be sent when it goes unanswered for a duration that matches the Timeout for the group.
|-
| style="text-align: center;" | '''Timeout''' 
| Defines, in seconds, the total time a call will spend in the ring group before being sent to the Failover Destination
|-
| style="text-align: center;" | '''Allow Recording''' 
| Determines if feature code recording is allowed for calls in this queue.
If YES = enables the recording feature.

If NO = disables the recording feature.

|-
| style="text-align: center;" | '''Agent Ring Time''' 
| Defines, in seconds, how long the queue will ring a particular agent. If using the Ring All strategy, this value should match the Timeout. For other strategies, this should be set to how long an extension should ring before moving on to the next.
|-
| style="text-align: center;" | '''Autofill''' 
|
This deals with how the queue handles multiple concurrent calls. If set to NO the queue will wait until the first call is answered before ringing phones with the second call. If set to YES the queue will ring phones for all calls as they come in.  Note this setting has a different effect with ring all that may not be desirable. 

When autofill is yes and ring strategy is 'ringall' the group will still deliver the calls in the order they are received and in order to prevent calls from being lost by phones that have too few lines for the number of calls in the queue, the group will only deliver a single additional call when Ring In Use is set to yes. If someone who's phone is ringing rejects the call, the group will deliver the next call in line to all members.  This means the the rejected call is no longer ringing the member's phone that rejected the call, however it is still ringing all of the other members.  At the same time the 2nd call in line comes in and rings everyone.

|-
| style="text-align: center;" | '''Ring In use''' 
| If '''YES''' = Distribute calls to group members if they are already on a call
If '''NO''' = Do NOT distribute calls to group members if they are already on a call

|}

The following outlines the steps to add/edit ring group settings.

'''STEPS:'''

#From the '''Destinations'''=>'''Group''' page, click the '''ADD Group''' button. The '''Edit Ring Group''' page appears.
#In the '''Name''' field, enter a name for the group.
#In the '''Group Number''' field, enter a number for the group. This number must be three or four digits in length.
#Check the '''Allow Paging with <nowiki>**+</nowiki> the Group Number''' to enable the setting.
#Select the desired '''Ring Strategy''' from the drop-down list.
#Select the desired '''Failover Destination''' from the list of available options.
#Enter the time for the '''Timeout''' parameter.
#Select “'''Yes'''” if you want to enable the '''Allow Recording '''setting. Otherwise, set it to “'''No'''”.
#Enter the time for the '''Agent Ring Time''' parameter.
#Select “'''Yes'''” if you want to enable the '''Autofill '''setting. Otherwise, set it to “'''No'''”
#Select “'''Yes'''” if you want to enable the '''Ring in use '''setting. Otherwise, set it to “'''No'''”
#Click [[File:savechanges.png]] the button.
#From the '''Members''' section of the '''Edit Ring Group''' page, select the extension number from the list of extensions that you want to add to the ring group. Click the '''ADD''' button.
#A second list appears next to the Members list displaying the selected extension. Repeat step 13 until all extensions that you want to add are on the list. Additionally, Shift or Control click functionality works for adding multiple extensions at once.
#To remove an item from the list of additions, select the extension number then click the '''REMOVE''' button.
#Define '''Priorities''' for the extensions, if needed. The scale ranges 0 thru 9 (0 is the highest priority).
#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.

==== Ring Group Advanced Settings ====
[[File:customcallerid.png|center]]
The following table describes the parameters (fields) on the '''Advanced''' '''Settings''' for Ring Groups page and their descriptions and recommended settings.

{| class="wikitable"
|-
| colspan="2" | <center>'''Sections/Fields'''</center>
| <center>'''Description'''</center>
|-
| colspan="3" | <center>'''Ring/Hold Settings'''</center>
|-
| style="text-align: center;" | '''Caller Hears'''
| colspan="2" | update text

|-
| style="text-align: center;" | '''Override Hold Music'''
| colspan="2" | update text
|-
|style="text-align: center;" | '''Hold Music'''
| colspan="2" | update text
|-
| style="text-align: center;" | '''Ring Tone'''
| colspan="2" | When Caller Ring Settings are set to Ringing, use this to define the ring cadence.
|-
| colspan="3" | <center>'''Custom Caller ID Settings'''</center>
|-
| style="text-align: center;" | '''Use'''
| colspan="2" | Defines what method, if any, you wish to use to override Inbound CID. The options are:
System Default, Name, Number, Name and Number

|-
| style="text-align: center;" | '''Name'''
| colspan="2" | This is the name of the caller that will be displayed when a call is received.
|-
| style="text-align: center;" | '''Number'''
| colspan="2" | This is the number of the caller that will be displayed when a call is received.
|-
| style="text-align: center;" | '''Prepend'''
| colspan="2" | Selecting Prepend will enter the text you put in the field, followed by the original CID. If not prepending, the override eliminates the original information.
|}

==== Edit Advanced Ring Group Settings ====
'''STEPS:'''

# Select '''Destinations=>Groups'''. The '''Ring Groups''' page appears.
# Select the Ring Group you want to edit by either select the [[File:penciledit.png]] icon to the right of the group name or click on the blue Group Name link.
# The '''Edit Ring Group''' page appears. Scroll down the page and click on the '''Advanced''' link. The '''Advanced Ring Group Settings''' section is displayed.
# Make the necessary changes to the settings then click button [[File:savechanges.png]] 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.

==== Edit Custom Caller ID Settings ====
'''STEPS:'''

# Select '''Destinations=>Groups'''. The '''Ring Groups''' page appears.
# Select the Ring Group you want to edit by either select the [[File:penciledit.png]] icon to the right of the group name or click on the blue Group Name link.
# The '''Edit Ring Group''' page appears. Scroll down the page and click on the '''Advanced''' link. '''The Custom Caller ID''' section is displayed.
# Make the necessary changes to the settings then click [[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.

==== Edit Members/Agents ====
[[File:editmembersagents.png|center]]
{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Members
| This is the list of available extensions that are in the database.
|-
| Agents
| This is the list of available agents that are in the database.
|-
| Priority
| This field allows you to set the priority level for the members or agents. The lower the number, the higher the priority.
|-
| ADD
| This button will add the select item on the list to the right of the available members or agents. Shift and Control click functionality works for adding more than one at a time.
|-
| REMOVE
| This button will remove the item(s) that have been selected from the list of available members or agents. Shift and Control click functionality works for adding more than one at a time.
|}

==== Add Agents/Members to the Group ====
'''STEPS:'''

# Navigate to the '''Destinations=>Groups=>Edit Group''' page.
# Select from the left hand column which agents/members to add to the group.
# Click the '''Add '''button.
# Click [[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 Agents/Members from the Group ====
'''STEPS:'''

# STEPS: Navigate to the '''Destinations'''=>'''Groups'''=>'''Edit Group '''page.
# Select from the right hand column which agents/members to remove from the group.
# Click the '''Add '''button.
# Click [[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

=== Automatic Call Distribution ([[ACD]]) ===

Systems licensed to use [[ACD]] will have access to additional functionality pertaining to Ring Groups. ACD allows for more flexibility and control over the queuing in a Ring Group, as well as allowing the creation of Agents who can be added to the Ring Group.

[[File:autocalldistpage.png]]

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Agent Retry Timer
| This defines, in seconds, how long the queue will wait before attempting to ring an agent again.
|-
| Weight
| The priority of the queue in relation to other queues in the system.
|-
| Wrap-up Time
| This defines, in seconds, how long the queue will wait after an agent has ended a call before flagging then as an active agent ready to accept more calls.
|-
| Autopause
| This will place an agent in a pause state in relation to this queue when they ignore or answer a call. To become an active member of the queue again unpause using the 1* feature code a phone or 1*+agent# for an agent.
|-
| Maximum # of people in queue
| Defines how many callers can join the queue before it is considered full.
|-
| Announce Frequency
| Must be set for Announce Holdtime to function. This pertains to how often the caller will hear the hold time announcement, in seconds
|-
| Periodic Announce Frequency
| This defines how much time passes between playing the recording that was set for this queue under Periodic Announcement.
|-
| Announce Holdtime
| If set to YES this will announce a caller’s status in the queue with regards to their expected hold time, pending the hold time is estimated to be over 1 minute. If the hold time is less than one minute, this will announce the caller’s position in the queue. A period of time is needed for the algorithm to ‘learn’ and estimate hold times accurately. This message will not play for the first caller in the queue. Caller Ring Settings must be set to a MOH file in order for this message to play.
|-
| Join Empty Queue
| Controls if a caller can join a queue.
YES = Callers can join the queue with no members or whose members are busy.

NO = Callers can only join the queue with active, non-busy members.

STRICT = Callers can join the queue as long as there are active members, be they available or busy.

|-
| Leave Empty Queue
| Controls if a caller will leave an empty queue.
YES = Callers will leave the queue if there are no active members, or all members are busy.

NO = Callers will not leave the queue even if there are no active members, or all members are busy.

STRICT = Callers will leave the queue if there are no active members, but will remain in the queue if all members are busy.

|-
| Report Hold Time
| Set to YES if you want the person answering the queue call to hear a message indicating how long the call they just answered was in the queue
|-
| Member Delay before connect
| Delay, in seconds, before a member is connected to the queue.
|-
| Timeout Restart
| If set to YES the ring group timeout will reset after attempting to ring an agent/member. When set to NO, time taken to ring an agent/member will be subtracted from the ring group time, and when the timer ends the group will send the caller to the Failover Destination
|-
| Service Level
| Set this to the time, in seconds, that defines the acceptable level of service for the queue. The Live Queue Data page will track the percentage of calls answered within the allotted time.
|-
| Exit Status: FULL
Exit Status: JOINEMPTY

Exit Status: LEAVEEMPTY

Exit Status: JOINUNAVAIL

Exit Status: LEAVEUNAVAIL

| These are the destinations where calls will go based on the exit strategy defined. If no one in the group accepts the call this is the destination to which the call will be sent. Destinations can be extensions, other ring groups, a menu or a voicemail box.
|-
| Intro Announcement
| Define what prompt to play to the caller just before they enter the queue. (Previously “Announce Filename”)
|-
| Agent Announcement
| Defines what prompt to play to the agent/member who answers the queue call. This setting is set per Queue and therefore can be used to identify what Queue is being answered (for agents handling calls from multiple Queues.
|-
| Periodic Announcement
| Defines what prompt to play to the caller at the Periodic Announce frequency interval. This will not play for the caller in the first position of the queue. Caller Ring Settings must be set to a MOH file in order for this message to play.
|-
| Exit Menu
| Set this to a menu. Key presses while in this group will send people the corresponding destination from that menu.
|}

''Table 39Edit Ring Group ACD Agents Settings and Descriptions''

==== Edit Automatic Call Distribution (ACD) Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Make the necessary updates to the ACD settings.
#Click [[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.

==== Add Automatic Call Distribution (ACD) Agents ====
[[File:addeditagentfeatures.png]]
{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Name
| Enter a name that will identify this agent.
|-
| ID
| Enter a unique 3-4 digit number to identify this agent. This number is used to log in/out.
|-
| PIN
| Enter a minimum of 4 digits for the PIN. This is the password used to log in/out with the agent.
|}

''Table 40Add/Edit Agent Features and Descriptions''

 '''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The Agents page appears.
#Click on the '''ADD''' button to add a new agent. A box with the name, ID and PIN field appears. Enter the information for the new agent.
#Click [[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.

==== Edit Automatic Call Distribution (ACD) Agent ====

'''STEPS:'''

#From the '''Destinations''' '''Groups Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The '''Agents''' page appears.
#Update the necessary information for agent(s) then click the '''SAVE CHANGES''' button. The changes that you made will appear in the updated list.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

==== Delete Automatic Call Distribution (ACD) Agent ====

'''STEPS:'''

#From the '''Destinations'''=>'''Groups=>Edit Group '''page, scroll to the '''ACD Settings''' section.
#Click on the blue '''Agents''' link to the right of the words '''ACD Settings'''. The '''Agents''' page appears.
#From the list of existing agents, click on the box (placing a checkmark) to the left of the Agent(s) that you want to delete. Click the '''DELETE SELECTED''' button.
#The agent selected is removed from the list.
#Click [[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.

=== Multicast Paging Group ===

A Multicast Paging Group provides a way to page a large number of phones, while keeping the burden on the network and PBX to a minimum.

'''NOTE: IPitomy recommends using multicast paging for groups of 10 or more phones that are paged frequently.'''
[[File:multicastgroups.png|center]]
[[File:editmultipagewindow.png|center]]
 The following table describes the fields (parameters) on the Multicast Edit Paging Group window and the recommended settings.

{| class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| Name
| This is a name of the multicast paging group.
|-
| Local Number
| This is the number that will be dialed to launch the multicast paging to the defined group. This is the unique 3-4 digit number.
|-
| Multicast Address
| This should be a valid multicast IP for a Multicast Address<ref name="ftn1">224.0.0.0 to 239.255.255.255, with many addresses reserved.</ref>. We recommend '''225.5.5.5'''. You can use the same IP for each group, so long as you assign them different ports.
|-
| Multicast Port
| This must be a valid, unused port Multicast Port.
'''NOTE: This must be a port below 6535.'''

|-
| Time to Live
| Time to Live is a counter that decreases by 1 each time a packet passes through a router. When the counter reaches 0, the packet is considered dead and will not pass to any other networks. This can be left blank or configured as needed for your particular install.
'''NOTE: Leaving it blank the packet will not pass to another network.'''

|-
| Members
| The Members section allows you to add the extensions to be paged within the group when a page is sent.
|-
| Add/Remove
| The Add and Remove button allows you to add or remove members (extensions) from the paging group.
|-
| Create/Update
| Depending on the process that is being performed, this button will appear (be labeled) as either Create or Update. This will create a new member or process the changes made.
|-
| Close Window
| This link will close the Edit Page Group window and return you to the Ring Groups page.
|}

The following section outlines the steps to add a new Multicast Paging Group.

==== Add Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate to the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on the '''Add Groups '''button. The '''Edit Page Group''' page appears.
#Enter the necessary information for the new paging group.
#From the '''Members''' section of the '''Edit Page Group''' window, select the members (extensions) that you want to include in the paging group.

Use the CTRL or SHIFT button to select multiple or a range of extensions from the list.

#Click on the '''ADD''' button to add the selected members to the group.
#Click the '''CREATE''' button to save the changes. The “Group Created” message appears. Close the window (click on the “X”). The system returns you to the '''Ring Groups''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones that correspond to extensions that were added to the '''Multicast Paging Group'''. Once all the phones have been re-booted, you will be able to utilize this feature by simply dialing the Local Number that was just created.

==== Edit Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on [[File:penciledit.png]] icon to the right of the '''Name '''of the paging group you want to update. The '''Edit Page Group''' page appears.
#Enter the necessary information for the paging group.
#Click the '''UPDATE''' button to save the changes. Close the window (click on the “X”). The system returns you to the '''Ring Groups''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones for all the extensions that were involved in the Multicast Paging Group.

==== Delete Multicast Paging Group ====

'''STEPS:'''

#Navigate to the '''Destinations'''=>'''Group '''page
#Locate the '''Multicast Paging Groups''' section (at the bottom of the page).
#Click on [[File:deleteselected.png]] icon to the right of the '''Name '''of the paging group you want to delete. The group is removed from the '''Multicast Paging Group''' page.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database
#'''Re-boot''' the phones for all the extensions that were involved in the Multicast Paging Group which was deleted.

=== Live Queue Data ===

The Live Queue Data page displays current call statistics for ACD queues. The page will refresh every 10 seconds, allowing a manager to monitor the activity in the queue as it happens. You can Apply Changes if you would like to reset the data.

[[File:ringlivequedata.png|center]]

{| style="border-spacing: 0px;" class="wikitable"
|-
| <center>Sections/Fields</center>
| <center>Description</center>
|-
| <queue selection box>
| In the queue selection box the Queue or Queues to be displayed can be selected. Use the Control key (Command on Mac) to select multiple queues for display.
|-
| <queue statistics box>
| The Queue Statistics box has been reformatted in SW release 3.4.1 to allow its use on call center wall boards (large PC displays).
Each queue is encapsulated with a thin-line box. Queue statistics are color-coded for quick orientation on wall boards. Since this data is presented in GUI format it can be easily resized to accommodate large displays.

The top line of the queue display presents the Queue Name and the date.

Statistics are a running total and will reset to zero whenever Apply Changes is clicked (as is the case of the screen capture above).

|-
| Time in Q
| This is the average time that callers have waited in queue before being answered by a queue member/agent.
|-
| Calls in Q
| This is the number of calls that are waiting in the queue.
|-
| Answered
| This is the number of calls answered by members/agent of this queue.
|-
| Abandoned
| This is the number of calls that have not been answered and the caller abandoned the queue.
|-
| Target Time
| Displays the service level set for this ACD queue. The Service Level is the arbitrary time allocated by the queue administrator which is set as a gauge by which all calls should be answered.
|-
|  % in Target
| This is the percentage of answered calls measured against the service level. (The percentage of calls being answered within the set Target Time.)
|-
| <Member/Agent field>
| At the right side of the Queue Statistics box the agents and queue members are listed. There are two maximum columns and the window will resize to accommodate all members and agents registered to this queue.
Member/Agent status is available at a glance with color-codes as follows:

{| style="border-spacing:0;"
|-
| style="background-color:#c00000;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Red'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | In Use
|-
| style="background-color:#006600;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Green'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Idle (Logged In)
|-
| style="background-color:#ffff00;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Yellow'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Ringing
|-
| style="background-color:#7f7f7f;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Gray'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Unavailable (Not Logged In)
|-
| style="background-color:#e36c0a;border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Orange'''</center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center><nowiki>=</nowiki></center>
| style="border:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Paused
|}

Also in the Member/Agent data are two numeral fields. The first number represents the Priority of this Member/Agent in this queue. The second number is the total calls taken by this Member/Agent.

|}

=== Live Queue Data Wallboard ===
Software version 3.4.1 and above allows the use of call center wall boards with Live Call Queue. This function is facilitated using a special user name and password via Smart Personal Console.

==== Activate Live Call Queue Wallboard ====
'''STEPS:'''

# Navigate to the '''PBX Setup'''=>'''General '''page
# Locate to the '''General Settings''' section (second section from the top).
# Locate the SPC Reports User Password field (bottom of section)[[File:spcreportsuserpass.png]]
# It may be helpful to change this to a user-friendly password. If you change the password be sure to [[File:savechanges.png]] and '''Apply Changes'''.
# Make a note of this password.
# On the PC that will be used for the wallboard display, open a web browser and navigate to the IPitomy PBX login screen
# Using the right sideSmart Personal Console (User Login) input the word “reports” as the user name and the password saved in step 3. [[File:reportslogin.png]]
# Click on '''Login''' .
# The CDR-Report page is opened. On this page is a button to select Live Queue.
# Click on '''Live Queue'''.
# The same display presented in PBX Administration is displayed. Simply select the group or groups that you wish to display on the wallboard.
# Size this web page data output for the monitor in use by pressing and holding the Cntrl key (PC, Command key on Mac) and press “+” or “-“ until the data is sized as you wish.

----
<references/>

IP PBX Manual Destinations Menus

2013-06-25T12:27:29Z

Nick Branica:

== Menus ==
Menus direct callers to destinations within a business. A menu can route a caller to a destination once a number on a key-pad has been pressed. The menu prompt tells a caller what number to press to get to a desired destination. Menus can also be used to provide information to callers like driving directions, etc.

*Menu training Video - [https://www.youtube.com/watch?v=XjRzyUEpOfI&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

[[File:menuspage.png]]
=== Edit Menu ===
[[File:editmenupage.png]]
{| style="border-spacing: 0px;" class="wikitable"
|-
| '''Sections/Fields'''
| colspan="2" | '''Description'''
|-
| colspan="2" style="text-align: center;" | '''Name'''
| Enter a name for the menu.
|-
| colspan="2" style="text-align: center;" | '''Menu Number (dial to go to menu)'''
| This is the number of the menu. This number is used to access the menu from extensionsif necessary. The number must be a unique 3 or 4 digit number.
|-
| colspan="2" style="text-align: center;" | '''Default Menu Prompt '''
| This is the recording that is active for callers who reach this menu when the Menu Prompt is set to default.
|-
| colspan="2" style="text-align: center;" | '''Active Menu Prompt'''
| This function is enhanced in version 3.4.1. (Previously “Menu Prompt”.) As it indicates, this selection determines what prompt recording is heard by callers who reach this menu.
Record this first under PBX SetupPrompts.

|-
| colspan="2" | '''Control Menu Prompt:''''''Copy To'''
| The Copy To function allows you to place pre-recorded (and previously uploaded, see: PBX SetupPrompts) prompts into the available Prompts to be associated to this menu. Available “slots” are 1-5 where the user may then select 0-5, 0 being the default and 1-5 being those that are loaded. The user also has the ability to rerecord the prompts 1-5 via DTMF Administration access.
Note: uploaded prompts are “copied” to the menu; therefore rerecorded prompts do not overwrite the original uploaded file.

|-
| colspan="2" style="text-align: center;" | '''Fail Over'''
| This is the destination that the menu will fail over to if the user does not press a digit in the allotted Response Time.
|-
| colspan="2" style="text-align: center;" | '''Override Hold Musc'''
| update text
|-
| colspan="2" style="text-align: center;" | '''Hold Music'''
| update text
|-
| colspan="3" style="border: 0.0069in solid rgb(0, 0, 255); padding: 0in 0.075in; text-align: center;" | '''Assign Menu Destinations'''
|-
| colspan="2" style=text-align: center;" | '''Menu Destinations '''
|This list represents the destinations callers are routed to when pressing 0-9,*, <nowiki>#, and FAX. </nowiki>If a destination is set for FAX, the PBX will detect if a call inbound over an Analog or T1/PRI card is a fax while the prompt is playing, and will route calls to this destination if fax tone is detected.
|}

==== Add Menu Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Menus '''page, click the '''ADD''' menu button. The '''Edit Menu''' page appears.
#Enter the information for the new Menu item and make necessary destination assignments.
#Click button the [[File:savechanges.png]]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

==== Edit Menu Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Menus '''page, locate the name of the Menu item you want to edit.
#Click on [[File:penciledit.png]] icon to the right of the '''Name '''of the menu you want to update. The '''Edit Menu''' page appears.
#Make the necessary updates to the menu setting then click [[File:savechanges.png]] button.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

==== Delete Menu Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Menus '''page, locate the name of the Menu item you want to edit.
#Click on {File:deleteselected.png]] icon to the right of the '''Name '''of the menu you want to delete. The Menu name is removed from the Menu listing.
#Click [[File:savechanges.png]] button.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database.

==== Advanced Menu Settings ====

{| style="border-spacing:0;"
|-
| style="background-color:#b8cce4;border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Sections/Fields'''</center>
| style="background-color:#b8cce4;border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Description'''</center>
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Response Timeout
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | This is the number of seconds (0-60) the PBX will wait for digits to be dialed. If nothing is dialed in the allotted time, the menu will route the call to the Failover Destination. The minimum and default value is '''4'''.
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Digit Timeout
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | While entering digits, the PBX will wait this many seconds (1-20) after the last digit was entered to take any actions. Single decimal digits are permitted. The default value is '''2'''.
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Play the Greeting
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | This is the number of times the greeting will be played before the call is sent to the fail over destination. Default and minimum value is '''1'''.
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Local extension dialing
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | This indicates whether you are able to dial local extensions from the menu. Default value is '''YES.'''
Note: It is not necessary to designate leading digits for extensions. This setting alone indicates when digits collected should be compared to the available list of extensions.

|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Prompt Padding
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | This is the time in seconds that the PBX will wait when entering the menu before playing the prompt. The default is set to '''2''' seconds.
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Enable Admin PIN
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | This enables the use of DTMF Menu Prompt Administration. When enabled the user may access the menu from a DTMF-enabled device and input the PIN to enter into Menu Prompt Administration Default value is '''No'''.
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Admin PIN
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | <nowiki>This is the PIN used to access DTMF Menu Prompt Administration. Default value is: <empty>.</nowiki>
|}

IP PBX Manual Destinations Menus

2013-06-25T12:27:11Z

Nick Branica:

== Menus ==
Menus direct callers to destinations within a business. A menu can route a caller to a destination once a number on a key-pad has been pressed. The menu prompt tells a caller what number to press to get to a desired destination. Menus can also be used to provide information to callers like driving directions, etc.

**Menu training Video - [https://www.youtube.com/watch?v=XjRzyUEpOfI&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

[[File:menuspage.png]]
=== Edit Menu ===
[[File:editmenupage.png]]
{| style="border-spacing: 0px;" class="wikitable"
|-
| '''Sections/Fields'''
| colspan="2" | '''Description'''
|-
| colspan="2" style="text-align: center;" | '''Name'''
| Enter a name for the menu.
|-
| colspan="2" style="text-align: center;" | '''Menu Number (dial to go to menu)'''
| This is the number of the menu. This number is used to access the menu from extensionsif necessary. The number must be a unique 3 or 4 digit number.
|-
| colspan="2" style="text-align: center;" | '''Default Menu Prompt '''
| This is the recording that is active for callers who reach this menu when the Menu Prompt is set to default.
|-
| colspan="2" style="text-align: center;" | '''Active Menu Prompt'''
| This function is enhanced in version 3.4.1. (Previously “Menu Prompt”.) As it indicates, this selection determines what prompt recording is heard by callers who reach this menu.
Record this first under PBX SetupPrompts.

|-
| colspan="2" | '''Control Menu Prompt:''''''Copy To'''
| The Copy To function allows you to place pre-recorded (and previously uploaded, see: PBX SetupPrompts) prompts into the available Prompts to be associated to this menu. Available “slots” are 1-5 where the user may then select 0-5, 0 being the default and 1-5 being those that are loaded. The user also has the ability to rerecord the prompts 1-5 via DTMF Administration access.
Note: uploaded prompts are “copied” to the menu; therefore rerecorded prompts do not overwrite the original uploaded file.

|-
| colspan="2" style="text-align: center;" | '''Fail Over'''
| This is the destination that the menu will fail over to if the user does not press a digit in the allotted Response Time.
|-
| colspan="2" style="text-align: center;" | '''Override Hold Musc'''
| update text
|-
| colspan="2" style="text-align: center;" | '''Hold Music'''
| update text
|-
| colspan="3" style="border: 0.0069in solid rgb(0, 0, 255); padding: 0in 0.075in; text-align: center;" | '''Assign Menu Destinations'''
|-
| colspan="2" style=text-align: center;" | '''Menu Destinations '''
|This list represents the destinations callers are routed to when pressing 0-9,*, <nowiki>#, and FAX. </nowiki>If a destination is set for FAX, the PBX will detect if a call inbound over an Analog or T1/PRI card is a fax while the prompt is playing, and will route calls to this destination if fax tone is detected.
|}

==== Add Menu Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Menus '''page, click the '''ADD''' menu button. The '''Edit Menu''' page appears.
#Enter the information for the new Menu item and make necessary destination assignments.
#Click button the [[File:savechanges.png]]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

==== Edit Menu Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Menus '''page, locate the name of the Menu item you want to edit.
#Click on [[File:penciledit.png]] icon to the right of the '''Name '''of the menu you want to update. The '''Edit Menu''' page appears.
#Make the necessary updates to the menu setting then click [[File:savechanges.png]] button.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database

==== Delete Menu Settings ====

'''STEPS:'''

#From the '''Destinations'''=>'''Menus '''page, locate the name of the Menu item you want to edit.
#Click on {File:deleteselected.png]] icon to the right of the '''Name '''of the menu you want to delete. The Menu name is removed from the Menu listing.
#Click [[File:savechanges.png]] button.
#Click the '''Apply Changes''' link located at the top right hand corner of the page, to commit the changes to the database.

==== Advanced Menu Settings ====

{| style="border-spacing:0;"
|-
| style="background-color:#b8cce4;border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Sections/Fields'''</center>
| style="background-color:#b8cce4;border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | <center>'''Description'''</center>
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Response Timeout
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | This is the number of seconds (0-60) the PBX will wait for digits to be dialed. If nothing is dialed in the allotted time, the menu will route the call to the Failover Destination. The minimum and default value is '''4'''.
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Digit Timeout
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | While entering digits, the PBX will wait this many seconds (1-20) after the last digit was entered to take any actions. Single decimal digits are permitted. The default value is '''2'''.
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Play the Greeting
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | This is the number of times the greeting will be played before the call is sent to the fail over destination. Default and minimum value is '''1'''.
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Local extension dialing
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | This indicates whether you are able to dial local extensions from the menu. Default value is '''YES.'''
Note: It is not necessary to designate leading digits for extensions. This setting alone indicates when digits collected should be compared to the available list of extensions.

|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Prompt Padding
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | This is the time in seconds that the PBX will wait when entering the menu before playing the prompt. The default is set to '''2''' seconds.
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Enable Admin PIN
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | This enables the use of DTMF Menu Prompt Administration. When enabled the user may access the menu from a DTMF-enabled device and input the PIN to enter into Menu Prompt Administration Default value is '''No'''.
|-
| style="border-top:0.0069in solid #0000ff;border-bottom:0.0069in solid #0000ff;border-left:0.0069in solid #0000ff;border-right:none;padding-top:0in;padding-bottom:0in;padding-left:0.075in;padding-right:0.075in;" | Admin PIN
| style="border:0.0069in solid #0000ff;padding-top:0in;padding-bottom:0in;padding-left:0.5in;padding-right:0.075in;" | <nowiki>This is the PIN used to access DTMF Menu Prompt Administration. Default value is: <empty>.</nowiki>
|}

Alerts

2013-06-25T12:22:39Z

Nick Branica:

Alerts can be set up to allow notifications to be sent out when a certain number is dialed as an outgoing call, such as 911. Alerts allow a large community to be informed about about the outbound dial quickly. These notifications can be sent as an email, text message, and as an audio tone or message over the phone system and paging system. The alerts can be configured to be sent to only certain extensions or to all extensions.

This alert notification can be sent out to cell phones as well as computers.
Link to Alerts video training [http://www.youtube.com/watch?v=I10Srs1bseE&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

{| border="1" cellspacing="1" cellpadding="1" style="width: 750px;"
|-
| style="text-align: center;" | [[File:Sms.png]] 
| [[File:Email.png|Email Alert message]] 
| [[File:Text.png]] 
|-
| style="text-align: center;" | '''Mobile Phone SMS Alert message'''
| style="text-align: center;" | '''Email Alert Message'''
| style="text-align: center;" | '''Text Message on Manager'''
|}

=== Create Alerts ===

[[File:Alertdetail.png|center|Alertdetail.png]]

{| class="wikitable"
|-
! Sections/Fields
! Description
|-
| colspan="2" | '''Alert General Settings'''
|-
| Alert Name
| Enter what the alert will be called
|-
| Enable/Disable this Alert
| Enabling this alert will allow for notifications to be sent when the desired number is dialed. Disabling this alert will not allow for this function. If you would like notifications to be sent out when a certain outbound number is dialed, be sure to have this feature enabled.
|-
| Enable/Disable Text Notifications for this Alert
| Enabling text notifications will allow chosen recipients to be alerted via their QManager text messages. Disabling this feature will not allow QManager notifications.
|-
| Enable/Disable Email Notifications for this Alert
| Enabling email notifications will allow emails to be sent out to chosen recipients when the alert is set off.
This feature can also send alerts to cell phones as a text message if the recipient has a free gateway service through their cell phone provider such as “@vtext.com” that Verizon provides. (How to send SMS Messages [[http://wiki.ipitomy.com/wiki/How_to_send_SMS_Messages [1]]])

Disabling this feature will not send alerts to email or cell phones.

|-
| Enable/Disable Scheduled Call Notifications for this Alert
| Enabling Scheduled Call notifications sends out an alert tone of your choice to the desired phone extensions. This feature does not require the recipient to be near their computer or phone to receive the alert notification.
Disabling this alert will not allow the tone to be sent out to desired extensions.

|-
| Next Destination to Route Caller (non-outbound)
| When using alerts as a destination, callers who have dialed a specific DID that is routed to the alert or pressed a single digit in a menu that is routed to an alert can combine sending a call and announcing the call with email, text, or external audio messaging.
An example of applying this feature would be in a retail store. If an employee dials for a certain department, there can be notifications sent out via email, text, or audio notifications through out the store alerting the next available employee in the desired department.

|-
| colspan="2" | '''Alert Text to be Displayed with this Notification'''
|-
| Email Subject Text
| Enter the desired alert text that will be sent out via email or text message as the subject heading
|-
| Email/Text Message Body
| Enter what will be sent out in the alert message to email addresses or text alerts.
%DNID: the number that has been dialed out %CALLERID: the extension ID that has dialed the number %TIME: the date and time the number was dialed

| colspan="2" | Client Text Alerts
|-
| Extensions
| All of the extensions that are part of your system are displayed here. Highlight the ones that you would like to send text alerts to over the QManager application and click add.
|-
| QManager Recipients
| The extensions that you see here are the extensions that will receive text alerts over the QManager application.
|-
| colspan="2" | '''Email Alerts'''
|-
| Email Address
| Enter the email address that you would like to receive email alerts for this notification and click add.
|-
| Email Recipients
| The addresses you see here are the recipients of the email notifications for this alert.
|-
| colspan="2" | '''Scheduled Call Alerts'''
|-
| Scheduled Calls
| The extensions that you see here are the extensions that will receive text alerts through an audio notification over their phone.
|-
| Applied to Alert
| The extensions that you see here are the extensions that will receive text alerts through an audio notification over their phone.
|}

 '''To create an alert:'''

#Navigate to the Alerts page under the Destinations tab.
#Click on Create New Alert.
#Enter the Alert Name.
#Enable the the Alert and any notification system you would like to use.
#Add any extensions you would like to be notified when this Alert is set off for text alerts in the QManager, email, and scheduled call alerts.
#Click on [[File:Savechanges.png]].

=== Edit Alerts ===

This will allow you to enter the outgoing number that will set off the alerts as well as the notification message you wish to go out with this alert. [[File:Editoutgoingroute.png|center|Editoutgoingroute.png]]

{| class="wikitable"
|-
! Sections/Fields
! Description
|-
| colspan="2" | '''Edit Outbound Route'''
|-
| Route Name
| The name of the outbound route for the alert
|-
| Route Type
| The type of outbound route that the alert is. For example, if the alert is for 911, then it would be an emergency outbound route.
|-
| Start Pattern
| This field allows you to enter the patter of digits that would first be entered to set off the alert system.
|-
| Digits
| This field allows you to enter the length of the number that would be dialed for the alert.
|-
| Exact Length
| This gives you a drop down menu that includes yes or no options. The number that is entered can be exactly number of digits that has been specified in the previous field or not.
|-
| Trunks
| See Outgoing Call Routing [[http://wiki.ipitomy.com/wiki/IP_PBX_Manual_Call_Routing#Outgoing_Call_Routing [2]]]
|-
| Assign Alert Notification
| This drop down menu will give you options for the alert notifications you have created when creating an alert. select the notification that is appropriate for the alert.
|}

#Under the Call Routing Tab, navigate to the Outgoing page.
#Click on the Outbound Route that you would like to edit.
#Enter the route name, start patter, digits, exact length, and select an alert notification for this alert.
#Click on [[File:Savechanges.png]].

Training:Application Solution

2013-06-25T12:19:59Z

Nick Branica:

= IPitomy Technical Training – Basic =

== Understanding the Application ==

 The application is the result of a properly designed PBX Solution that meets the requirements of the customer. The most effective way of accomplishing the customer specific application is to get all of the information from the customer and use that information to populate your IPitomy setup worksheet. In the preceding modules we’ve covered the preliminary ground work needed to ask the questions that will allow you to build the application.  The setup worksheet is a document that should define the scope of work as far as programming the system.  Having this in writing provides a clear application definition and reduces the possibility of misalignment of expectations.

The IPitomy worksheet is an excellent place to start this process and provides the written backup that is very often a needed fail-safe plan to reconstruct the application in the absence of a good backup.  It is a good idea to keep a copy of the setup worksheet in the same directory that you use to store the customers backup files.  This module covers the basic system application.

= Database Building: =

IPitomy is an IP PBX system which means is is a pure software based business telecommunication system.  It is very similar to traditional TDM based key systems and PBX's with one big difference; it is a pure software based system. IPitomy does not have all of the hardware components associated with traditional systems so there are very few parts that can fail. Building the database is best done when well planned. Planning assures that the solution meets the expectations of the customer. Obtaining the information prior to the date of installation provides you with the data you need to install professionally and with the least amount of un-billable call backs.

IPitomy has a web based administration interface that makes programming and setup of the application simple.  IPitomy does not come with a default database, so it is necessary to create the database in the proper sequence to build the elements necessary for successful call flow. 

IPitomy has a very simple setup process. 

*'''Destinations''' - provide a destination within the PBX to route calls to like an automated attendant or an extension. DID's can be used to directly ring any destination e.g. conference rooms or extensions.
*'''Providers''' are the elements that route calls into the system and out of the system (Trunks).  Once you set up the destinations, you can set up the providers and implement the design requirements for call routing.

[[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]]

=== Call Groups/ACD ===

*Call Groups Defined – Call Groups are a powerful communications resource in the IPitomy IP PBX and for your customer. Since there is no limit to the number of call groups possible in IPitomy, the use is unrestricted. ACD will be discussed in the Advanced training course.<ul style="list-style-type:circle;">
</ul></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 Call Groups (ACD Feature – callers in an ACD - Call 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 Call 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: 

=== DCM (Desktop Call Manager) ===

*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]

Training:Application Solution

2013-06-25T12:15:39Z

Nick Branica:

= IPitomy Technical Training – Basic =

== Understanding the Application ==

 The application is the result of a properly designed PBX Solution that meets the requirements of the customer. The most effective way of accomplishing the customer specific application is to get all of the information from the customer and use that information to populate your IPitomy setup worksheet. In the preceding modules we’ve covered the preliminary ground work needed to ask the questions that will allow you to build the application.  The setup worksheet is a document that should define the scope of work as far as programming the system.  Having this in writing provides a clear application definition and reduces the possibility of misalignment of expectations.

The IPitomy worksheet is an excellent place to start this process and provides the written backup that is very often a needed fail-safe plan to reconstruct the application in the absence of a good backup.  It is a good idea to keep a copy of the setup worksheet in the same directory that you use to store the customers backup files.  This module covers the basic system application.

= Database Building: =

IPitomy is an IP PBX system which means is is a pure software based business telecommunication system.  It is very similar to traditional TDM based key systems and PBX's with one big difference; it is a pure software based system. IPitomy does not have all of the hardware components associated with traditional systems so there are very few parts that can fail. Building the database is best done when well planned. Planning assures that the solution meets the expectations of the customer. Obtaining the information prior to the date of installation provides you with the data you need to install professionally and with the least amount of un-billable call backs.

IPitomy has a web based administration interface that makes programming and setup of the application simple.  IPitomy does not come with a default database, so it is necessary to create the database in the proper sequence to build the elements necessary for successful call flow. 

IPitomy has a very simple setup process. 

*'''Destinations''' - provide a destination within the PBX to route calls to like an automated attendant or an extension. DID's can be used to directly ring any destination e.g. conference rooms or extensions.
*'''Providers''' are the elements that route calls into the system and out of the system (Trunks).  Once you set up the destinations, you can set up the providers and implement the design requirements for call routing.

[[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]]

=== Call Groups/ACD ===

*Call Groups Defined – Call Groups are a powerful communications resource in the IPitomy IP PBX and for your customer. Since there is no limit to the number of call groups possible in IPitomy, the use is unrestricted. ACD will be discussed in the Advanced training course.<ul style="list-style-type:circle;">
</ul></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 Call Groups (ACD Feature – callers in an ACD - Call 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 Call Groups and ACD.

=== 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: 

=== DCM (Desktop Call Manager) ===

*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]

Training: Alerts

2013-06-25T09:32:51Z

Nick Branica:

Alerts are destinations where calls can be routed to trigger a notification alarm. Alerts can be used on inbound calls and outbound calls. On outbound calls, Alerts can send a text message, send an email and invoke a scheduled call; which will can trigger an audible alarm. For instance, if a caller dials 911, the system can be configured to send a text or email to a group of recipients who are designated to respond in emergencies. The email and text message can be configured to include the number dialed, the extension dialing the number and the date and time. By using the Scheduled Calls feature, an audio announcement can be triggered to play a prerecorded audio announcement over the paging system.

Training Video:
[http://www.youtube.com/watch?v=I10Srs1bseE&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

Training: Alerts

2013-06-25T09:31:28Z

Nick Branica: Created page with "Alerts are destinations where calls can be routed to trigger a notification alarm. Alerts can be used on inbound calls and outbound calls. On outbound calls, Alerts can send..."

Alerts are destinations where calls can be routed to trigger a notification alarm. Alerts can be used on inbound calls and outbound calls. On outbound calls, Alerts can send a text message, send an email and invoke a scheduled call will can trigger an audible alarm. For instance, if a caller dials 911, the system can be configured to send a text or email to a group of recipients who are designated to respond in emergencies. The email and text message can be configured to include the number dialed, the extension dialing the number and the date and time. By using the Scheduled Calls feature, an audio announcement can be triggered to play a prerecorded audio announcement over the paging system.

Training Video:
[http://www.youtube.com/watch?v=I10Srs1bseE&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

Training: Menus

2013-06-25T09:23:31Z

Nick Branica: Training video for menus

Menus are a destination where calls can be routed. Once routed to the menu, the menu can play a prompt, respond to single digit dialing digits by routing the call to a predetermined destination, allow extension dialing and provide access to the voice mail system and forwarding gateway. in addition to the dialing and routing, the Menu can also be used to "Tag" a call for a specific Music on Hold message or music for the entire duration of the call.

Training Video:
[https://www.youtube.com/watch?v=XjRzyUEpOfI&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

Training: Ring Groups

2013-06-25T09:12:58Z

Nick Branica: Ring Group training video.

Ring groups are destinations where a single call can be routed and one to many extensions can ring simultaneously.
Training Video:
[https://www.youtube.com/watch?v=dZavlZJW-18&feature=c4-overview&list=UU1cwHaNWU97sdZm3yEBqNeg]

Alerts

2013-06-04T01:03:38Z

Nick Branica:

Alerts can be set up to allow notifications to be sent out when a certain number is dialed as an outgoing call, such as 911. Alerts allow a large community to be informed about about the outbound dial quickly. These notifications can be sent as an email, text message, and as an audio tone or message over the phone system and paging system. The alerts can be configured to be sent to only certain extensions or to all extensions.

This alert notification can be sent out to cell phones as well as computers.

{| border="1" cellspacing="1" cellpadding="1" style="width: 750px;"
|-
| style="text-align: center;" | [[File:Sms.png]] 
| [[File:Email.png|Email Alert message]] 
| [[File:Text.png]] 
|-
| style="text-align: center;" | '''Mobile Phone SMS Alert message'''
| style="text-align: center;" | '''Email Alert Message'''
| style="text-align: center;" | '''Text Message on Manager'''
|}

=== Create Alerts ===

[[File:Alertdetail.png|center|Alertdetail.png]]

{| class="wikitable"
|-
! Sections/Fields
! Description
|-
| colspan="2" | '''Alert General Settings'''
|-
| Alert Name
| Enter what the alert will be called
|-
| Enable/Disable this Alert
| Enabling this alert will allow for notifications to be sent when the desired number is dialed. Disabling this alert will not allow for this function. If you would like notifications to be sent out when a certain outbound number is dialed, be sure to have this feature enabled.
|-
| Enable/Disable Text Notifications for this Alert
| Enabling text notifications will allow chosen recipients to be alerted via their QManager text messages. Disabling this feature will not allow QManager notifications.
|-
| Enable/Disable Email Notifications for this Alert
| Enabling email notifications will allow emails to be sent out to chosen recipients when the alert is set off.
This feature can also send alerts to cell phones as a text message if the recipient has a free gateway service through their cell phone provider such as “@vtext.com” that Verizon provides. (How to send SMS Messages [[http://wiki.ipitomy.com/wiki/How_to_send_SMS_Messages [1]]])

Disabling this feature will not send alerts to email or cell phones.

|-
| Enable/Disable Scheduled Call Notifications for this Alert
| Enabling Scheduled Call notifications sends out an alert tone of your choice to the desired phone extensions. This feature does not require the recipient to be near their computer or phone to receive the alert notification.
Disabling this alert will not allow the tone to be sent out to desired extensions.

|-
| Next Destination to Route Caller (non-outbound)
| When using alerts as a destination, callers who have dialed a specific DID that is routed to the alert or pressed a single digit in a menu that is routed to an alert can combine sending a call and announcing the call with email, text, or external audio messaging.
An example of applying this feature would be in a retail store. If an employee dials for a certain department, there can be notifications sent out via email, text, or audio notifications through out the store alerting the next available employee in the desired department.

|-
| colspan="2" | '''Alert Text to be Displayed with this Notification'''
|-
| Email Subject Text
| Enter the desired alert text that will be sent out via email or text message as the subject heading
|-
| Email/Text Message Body
| Enter what will be sent out in the alert message to email addresses or text alerts.
%DNID: the number that has been dialed out %CALLERID: the extension ID that has dialed the number %TIME: the date and time the number was dialed

| colspan="2" | Client Text Alerts
|-
| Extensions
| All of the extensions that are part of your system are displayed here. Highlight the ones that you would like to send text alerts to over the QManager application and click add.
|-
| QManager Recipients
| The extensions that you see here are the extensions that will receive text alerts over the QManager application.
|-
| colspan="2" | '''Email Alerts'''
|-
| Email Address
| Enter the email address that you would like to receive email alerts for this notification and click add.
|-
| Email Recipients
| The addresses you see here are the recipients of the email notifications for this alert.
|-
| colspan="2" | '''Scheduled Call Alerts'''
|-
| Scheduled Calls
| The extensions that you see here are the extensions that will receive text alerts through an audio notification over their phone.
|-
| Applied to Alert
| The extensions that you see here are the extensions that will receive text alerts through an audio notification over their phone.
|}

 '''To create an alert:'''

#Navigate to the Alerts page under the Destinations tab.
#Click on Create New Alert.
#Enter the Alert Name.
#Enable the the Alert and any notification system you would like to use.
#Add any extensions you would like to be notified when this Alert is set off for text alerts in the QManager, email, and scheduled call alerts.
#Click on [[File:Savechanges.png]].

=== Edit Alerts ===

This will allow you to enter the outgoing number that will set off the alerts as well as the notification message you wish to go out with this alert. [[File:Editoutgoingroute.png|center|Editoutgoingroute.png]]

{| class="wikitable"
|-
! Sections/Fields
! Description
|-
| colspan="2" | '''Edit Outbound Route'''
|-
| Route Name
| The name of the outbound route for the alert
|-
| Route Type
| The type of outbound route that the alert is. For example, if the alert is for 911, then it would be an emergency outbound route.
|-
| Start Pattern
| This field allows you to enter the patter of digits that would first be entered to set off the alert system.
|-
| Digits
| This field allows you to enter the length of the number that would be dialed for the alert.
|-
| Exact Length
| This gives you a drop down menu that includes yes or no options. The number that is entered can be exactly number of digits that has been specified in the previous field or not.
|-
| Trunks
| See Outgoing Call Routing [[http://wiki.ipitomy.com/wiki/IP_PBX_Manual_Call_Routing#Outgoing_Call_Routing [2]]]
|-
| Assign Alert Notification
| This drop down menu will give you options for the alert notifications you have created when creating an alert. select the notification that is appropriate for the alert.
|}

#Under the Call Routing Tab, navigate to the Outgoing page.
#Click on the Outbound Route that you would like to edit.
#Enter the route name, start patter, digits, exact length, and select an alert notification for this alert.
#Click on [[File:Savechanges.png]].

File:Text.png

2013-06-04T00:47:12Z

Nick Branica: -> Creation failed: Systemcheck showed missing or incorrect configurated, dependent commands!

File:Email.png

2013-06-04T00:44:46Z

Nick Branica: Email message for alerts -> Creation failed: Systemcheck showed missing or incorrect configurated, dependent commands!

Email message for alerts

Call Recording

2013-06-04T00:32:56Z

Nick Branica:

This feature allows recording of inbound and outbound calls. Extensions can be put into groups which makes it easier to keep track of calls. There can be jobs that only record inbound calls, outbound calls, or both.

There are currently 3 main mechanisms for recording calls on an IPitomy IPPBX System.

#Manual Recordings:   
#*Initiated by agent, or Extension User while on call.
#**These recordings are either in Agent Recordings (if they are recordings by an agent taking a queue call).  Or they are available in the work folder in the recording extension's mailbox. 
#*Initiated by QManager
#**These recordings are available through View Recordings in QManager
#IPitomy Call Recording Application
#*Licensed IPitomy Feature that records all calls on designated destinations.
#Third Party Recording
#*IPitomy has implemented integration with Trivium to support their call recording application.

[[File:callrecording.png|center]]

=== Create New Recording Job ===

#Navigate to the Call Recording Page under the Applications tab.
#Click Add Recording
#Enter the name of the recording group.
#Add Ring Groups to the recording job that are applicable to the group that you want to be recorded.
#Add Available Routes that you would like to be recorded.
#Click on the [[File:Savechanges.png]] button.

[[File:Editrecordingjob.png|center|Editrecordingjob.png]]

{| class="wikitable"
|-
! Sections/Fields
! Description
|-
| colspan="2" | '''Recording Settings'''
|-
| Name
| Enter the name of the Recording Job
|-
| Enable
| Option to enable or disable call recording
|-
| Maximum number of recordings
| Enter the maximum amount of calls you would like to have recorded. If the maximum number has been reached, then the old recordings will be deleted to make room for new ones.
|-
| Maximum age of recordings
| Enter the maximum age of recordings. This will delete recordings after a certain period of time. This will clear out old recordings automatically to make room for new recordings.
|-
| Available Ring Groups
| These are the groups that can be selected to include in the recording job
|-
| Selected
| These are the groups that have been added to be included in the recording job. These are the extensions that you will find recordings for when checking this recording job
|-
| Available Routes
| Routes that can be selected to be included in the recording job. When one of these routes is dialed by one of the ring groups selected the call will be recorded.
|-
| Selected
| These are the routes that have been selected to be recorded when dialed by the selected ring groups.
|}

===Viewing Recording Jobs===

[[File:managerecordingfiles.png|center]]

To View the recording jobs and manage or listen to the recorded calls:
#Navigate to the Call Recording page under the Applications tab.
#Click the [[File:viewjobs.png]] icon to view the recording files for the group you wish to view.
#To download/listen to the recording files, click the [[File:downloadbutton.jpg]] button.

Call Recording

2013-06-04T00:30:52Z

Nick Branica:

This feature allows recording of inbound and outbound calls. Extensions can be put into groups which makes it easier to keep track of calls. There can be jobs that only record inbound calls, outbound calls, or both.

There are currently 3 main mechanisms for recording calls on an IPitomy IPPBX System.

#Manual Recordings:   
#*Initiated by agent, or Extension User while on call.
#**These recordings are either in Agent Recordings (if they are recordings by an agent taking a queue call).  Or they are available in the work folder in the recording extension's mailbox. 
#*Initiated by QManager
#**These recordings are available through View Recordings in QManager
#IPitomy Call Recording Application
#*Licensed IPitomy Feature that records all calls on designated destinations.
#Third Party Recording
#*IPitomy has implemented integration with Trivium to support their call recording application.

[[File:callrecording.png|center]]

=== Create New Recording Job ===

#Navigate to the Call Recording Page under the Applications tab.
#Click Add Recording
#Enter the name of the recording group.
#Add Ring Groups to the recording job that are applicable to the group that you want to be recorded.
#Add Available Routes that you would like to be recorded.
#Click on the [[File:Savechanges.png]] button.

[[File:Editrecordingjob.png|center|Editrecordingjob.png]]

{| class="wikitable"
|-
! Sections/Fields
! Description
|-
| colspan="2" | '''Recording Settings'''|align="center"
|-
| Name
| Enter the name of the Recording Job
|-
| Enable
| Option to enable or disable call recording
|-
| Maximum number of recordings
| Enter the maximum amount of calls you would like to have recorded. If the maximum number has been reached, then the old recordings will be deleted to make room for new ones.
|-
| Maximum age of recordings
| Enter the maximum age of recordings. This will delete recordings after a certain period of time. This will clear out old recordings automatically to make room for new recordings.
|-
| Available Ring Groups
| These are the groups that can be selected to include in the recording job
|-
| Selected
| These are the groups that have been added to be included in the recording job. These are the extensions that you will find recordings for when checking this recording job
|-
| Available Routes
| Routes that can be selected to be included in the recording job. When one of these routes is dialed by one of the ring groups selected the call will be recorded.
|-
| Selected
| These are the routes that have been selected to be recorded when dialed by the selected ring groups.
|}

===Viewing Recording Jobs===

[[File:managerecordingfiles.png|center]]

To View the recording jobs and manage or listen to the recorded calls:
#Navigate to the Call Recording page under the Applications tab.
#Click the [[File:viewjobs.png]] icon to view the recording files for the group you wish to view.
#To download/listen to the recording files, click the [[File:downloadbutton.jpg]] button.

File:T38G Setup Guide.pdf

2013-03-01T20:26:59Z

Nick Branica:

{{RMImage
|Filename=File:T38G Setup Guide.pdf
|CopyrightBy=
|CreationDate=2013/03/01
|Uploader=Nick Branica
|UploadDate=2013/03/01
|Description=
|Keywords=
|Format=application/pdf
|RelatedArticles=User:Nick Branica
}}

File:T38G Setup Guide.pdf

2013-03-01T20:26:59Z

Nick Branica: uploaded a new version of "File:T38G Setup Guide.pdf"

How to setup the TG38
{{FileIndex |index=t38 programming guide this designed assist t38g telephones work with ipitomy pbx system the phones has many advanced features get basic enabled integrate most popular used system. account setup voice mail call pickup blf connecting telephone connect rj45 plug back phone marked âinternetâ lan can attached other jack âpcâ once unit connected check network status using soft keys select now write down address and use browse web interface t38g. admin steps tab enable enter label for line key display name extension number sip password confirm move next page set time parameters server phone. format backlight options upload logo brands http /www.ipitomy.com/images/stories/ipphones/logotag.png memory are programmable buttons right side all tg38 including sift ext expansion module park speed dial pause abc<extension number> abc2192 recommends removing dnd replacing pause. change }}

File:T38G Setup Guide.pdf

2013-02-11T22:18:29Z

Nick Branica: How to setup the TG38 -> Creation complete

SMDR

2013-01-22T21:10:17Z

Nick Branica:

To download the pdf go here:

[http://ipitomy.com/webrelease/IPitomy/IP1100+/SMDR.pdf http://ipitomy.com/webrelease/IPitomy/IP1100+/SMDR.pdf]

'''SMDR data from server'''

                The SMDR data tis outputted from the Reporting->Reports Page. (the export button will only give you a CDR-summary).                 The file should be polled by directly by calling [ip address]/ippbx/exportCSV.php .

 By default it will give you all completed calls that day, in SMDR format.

 

                To specify a range of times, you can use the get variables ‘etime’ and ‘stime’. If you provide an stime, it will give you all records from that time until the current time. If you specify both, it will give you records between those times.

 

Example:

[ip address]/ippbx/exportCSV.php?stime=[epoch]

 

 

'''The SMDR specification'''

 

{| style="width:771px" align="center" border="0" cellpadding="0" cellspacing="0" width="771"
|-
| style="width:99px; height:17px" nowrap="nowrap" |
== Start Time ==

| style="width:99px; height:17px" nowrap="nowrap" |
== End Time ==

| style="width:77px; height:17px" nowrap="nowrap" |
== Direction ==

| style="width:74px; height:17px" nowrap="nowrap" |
== Trunk ==

| style="width:130px; height:17px" |
== Call ID ==

| style="width:99px; height:17px" nowrap="nowrap" |
== Source ==

| style="width:99px; height:17px" nowrap="nowrap" |
== Source CID ==

| style="width:93px; height:17px" nowrap="nowrap" |
== Source Channel ==

|-
| style="width:99px; height:17px" nowrap="nowrap" |
1264086258

| style="width:99px; height:17px" nowrap="nowrap" |
1264086296

| style="width:77px; height:17px" nowrap="nowrap" |
Incoming

| style="width:74px; height:17px" nowrap="nowrap" |
DAHDI/1-1

| style="width:130px; height:17px" |
1264086247.6308

| style="width:99px; height:17px" nowrap="nowrap" |
7145554445

| style="width:99px; height:17px" nowrap="nowrap" |
7145554445

| style="width:93px; height:17px" nowrap="nowrap" |
DAHDI/1-1

|}

 

{| style="width:675px" align="center" border="0" cellpadding="0" cellspacing="0" width="675"
|-
| style="width:129px; height:32px" nowrap="nowrap" |
== Destination ==

| style="width:122px; height:32px" nowrap="nowrap" |
== Destination CID ==

| style="width:171px; height:32px" nowrap="nowrap" |
== Destination Channel ==

| style="width:99px; height:32px" nowrap="nowrap" |
== DID ==

| style="width:153px; height:32px" nowrap="nowrap" |
== Ring Group ==

|-
| style="width:129px; height:32px" nowrap="nowrap" |
Elaine Blodgett

| style="width:122px; height:32px" nowrap="nowrap" |
 

| style="width:171px; height:32px" nowrap="nowrap" |
SIP/2208

| style="width:99px; height:32px" nowrap="nowrap" |
9413062200

| style="width:153px; height:32px" nowrap="nowrap" |
rg_33

|}

{| style="width:679px" align="left" border="0" cellpadding="0" cellspacing="0" width="679"
|-
| style="width:109px; height:24px" |
== Hold Time ==

| style="width:84px; height:24px" nowrap="nowrap" |
== Status ==

| style="width:123px; height:24px" nowrap="nowrap" |
== Total Call Length ==

| style="width:92px; height:24px" |
== Final Entry ==

| style="width:115px; height:24px" |
== Account Code ==

| style="width:155px; height:24px" |
== Authorization Code ==

|-
| style="width:109px; height:19px" |
3

| style="width:84px; height:19px" nowrap="nowrap" |
TRANSFERRED

| style="width:123px; height:19px" nowrap="nowrap" |
35

| style="width:92px; height:19px" |
 

| style="width:115px; height:19px" |
 

| style="width:155px; height:19px" |
 

|}

 

 

 

 

'''Field Specification'''

 

Start Time to End Time is the total length of the call.

                                Epoch Time

                Direction is the direction the call in coming

Incoming, Outgoing, or Internal

 

                Trunk is the associated inbound/outbound trunk

                                (SIP/ or DAHDI/ or IAX/)

                Call ID

                                The call id of the parent call entry (ties multiple legs of a call together

                Source

                                The Nickname the PBX gives the Source      

                Source Caller ID

                                The Caller ID of the Source

                Source Channel

                                The Channel of the Source caller

                Destination          

                                The Nickname the PBX gives the Destination

Destination Caller ID

                                The Caller ID of the Destination

                Destination Channel

                                The Channel of the Destination caller

                DID

                                The DID the Call came in on (if applicable)

               

Ring Group Name

                If it is sent into a ring group, the name of that ring group.

Hold Time

                                The amount of time the party was waiting for an Agent

Status

*Abandoned: Call entered a Queue, but the caller hung up before the call was answered or met the failover requirement.
*ChanUnavail: No channel was able to be created for the call due to a possible error state for the intended destination.
*Agent Completed: Connected Queue call was terminated at the extension.
*Hangup: Call to a Menu, but the caller hung up before taking any actions.
*Caller Completed: Connected Queue call was terminated at the calling parties end.
*Cancel: Call from an Extension, but the caller hangs up before the call is connected.
*Answer: A connected call.
*No Answer: A call that met the ring time of an extension without being answered.
*Timeout Exit: Call to a Queue that meets the Timeout.
*VM Not Left: Call ended with the caller being sent to voicemail where they did not leave a message.
*VM Left: Call ended with the caller being sent to voicemail where they left a message.
*Transferred: Connected call successfully transferred.
*Busy: Call out a trunk returned busy status, or extension was busy.
*- : Defines the call is still active, and the status will update when completed. Click on the call record for details.
*Exited Empty Queue: Call to Queue that follows Exit Empty rule.
*Exited With Key: Call to Queue that follows the Exit Menu key press.
*Account Code: Reserved for future account name information
*Authorization Code: Reserved for future authorization code information

 

         

         

 

 

Example:

The call is incoming on a hardware channel 1 group 1, the call ID links the two records together. The call first goes to a ring group ‘rg_33’ is answered by SIP/2208 (Elaine), and was transferred after 35 seconds of talking. Because the final flag is not set, we know there will be more about this call in the future.

 

The second entry for the call was picked up by SIP/2210 (Chris) and lasted 81 seconds long. The final entry flag has a ‘1’ signifying it is the last leg of this call.

 

1264086258, 1264086296, Incoming, DAHDI/1-1, 1264086247.6308, 7145554445, 7145554445, Zap/1-1, Elaine Blodgett, , SIP/2208, 9413062200, rg_33, 3, TRANSFERRED,35,

 

1264086296, 1264086395, Incoming, DAHDI/1-1, 1264086247.6308, 7145554445, 7145554445, Zap/1-1, Chris Beavers, 2210, SIP/2210-087b12f0, 9413062200, , , ANSWER, 81, 1

SMDR

2013-01-22T21:07:13Z

Nick Branica:

To download the pdf go here:

[http://ipitomy.com/webrelease/IPitomy/IP1100+/SMDR.pdf http://ipitomy.com/webrelease/IPitomy/IP1100+/SMDR.pdf]

 

{| border="0" cellpadding="0" cellspacing="0" height="34" width="13"
|-
| style="width:166px" | 
| style="width:166px" | 
| style="width:166px" | 
| style="width:166px" | 
|-
| style="width:166px" | 
| style="width:166px" | 
| style="width:166px" | 
| style="width:166px" | 
|-
| style="width:166px" | 
| style="width:166px" | 
| style="width:166px" | 
| style="width:166px" | 
|-
| style="width:166px" | 
| style="width:166px" | 
| style="width:166px" | 
| style="width:166px" | 
|}

'''SMDR data from server'''

                The SMDR data tis outputted from the Reporting->Reports Page. (the export button will only give you a CDR-summary). The file should be polled by directly by calling [ip address]/ippbx/exportCSV.php .

 By default it will give you all completed calls that day, in SMDR format.

 

                To specify a range of times, you can use the get variables ‘etime’ and ‘stime’. If you provide an stime, it will give you all records from that time until the current time. If you specify both, it will give you records between those times.

 

Example:

[ip address]/ippbx/exportCSV.php?stime=[epoch]

 

 

'''The SMDR specification'''

 

{| style="width:771px" align="center" border="0" cellpadding="0" cellspacing="0" width="771"
|-
| style="width:99px; height:17px" nowrap="nowrap" |
== Start Time ==

| style="width:99px; height:17px" nowrap="nowrap" |
== End Time ==

| style="width:77px; height:17px" nowrap="nowrap" |
== Direction ==

| style="width:74px; height:17px" nowrap="nowrap" |
== Trunk ==

| style="width:130px; height:17px" |
== Call ID ==

| style="width:99px; height:17px" nowrap="nowrap" |
== Source ==

| style="width:99px; height:17px" nowrap="nowrap" |
== Source CID ==

| style="width:93px; height:17px" nowrap="nowrap" |
== Source Channel ==

|-
| style="width:99px; height:17px" nowrap="nowrap" |
1264086258

| style="width:99px; height:17px" nowrap="nowrap" |
1264086296

| style="width:77px; height:17px" nowrap="nowrap" |
Incoming

| style="width:74px; height:17px" nowrap="nowrap" |
DAHDI/1-1

| style="width:130px; height:17px" |
1264086247.6308

| style="width:99px; height:17px" nowrap="nowrap" |
7145554445

| style="width:99px; height:17px" nowrap="nowrap" |
7145554445

| style="width:93px; height:17px" nowrap="nowrap" |
DAHDI/1-1

|}

 

{| style="width:675px" align="center" border="0" cellpadding="0" cellspacing="0" width="675"
|-
| style="width:129px; height:32px" nowrap="nowrap" |
== Destination ==

| style="width:122px; height:32px" nowrap="nowrap" |
== Destination CID ==

| style="width:171px; height:32px" nowrap="nowrap" |
== Destination Channel ==

| style="width:99px; height:32px" nowrap="nowrap" |
== DID ==

| style="width:153px; height:32px" nowrap="nowrap" |
== Ring Group ==

|-
| style="width:129px; height:32px" nowrap="nowrap" |
Elaine Blodgett

| style="width:122px; height:32px" nowrap="nowrap" |
 

| style="width:171px; height:32px" nowrap="nowrap" |
SIP/2208

| style="width:99px; height:32px" nowrap="nowrap" |
9413062200

| style="width:153px; height:32px" nowrap="nowrap" |
rg_33

|}

{| style="width:679px" align="left" border="0" cellpadding="0" cellspacing="0" width="679"
|-
| style="width:109px; height:24px" |
== Hold Time ==

| style="width:84px; height:24px" nowrap="nowrap" |
== Status ==

| style="width:123px; height:24px" nowrap="nowrap" |
== Total Call Length ==

| style="width:92px; height:24px" |
== Final Entry ==

| style="width:115px; height:24px" |
== Account Code ==

| style="width:155px; height:24px" |
== Authorization Code ==

|-
| style="width:109px; height:19px" |
3

| style="width:84px; height:19px" nowrap="nowrap" |
TRANSFERRED

| style="width:123px; height:19px" nowrap="nowrap" |
35

| style="width:92px; height:19px" |
 

| style="width:115px; height:19px" |
 

| style="width:155px; height:19px" |
 

|}

 

 

 

 

'''Field Specification'''

 

Start Time to End Time is the total length of the call.

                                Epoch Time

                Direction is the direction the call in coming

Incoming, Outgoing, or Internal

 

                Trunk is the associated inbound/outbound trunk

                                (SIP/ or DAHDI/ or IAX/)

                Call ID

                                The call id of the parent call entry (ties multiple legs of a call together

                Source

                                The Nickname the PBX gives the Source      

                Source Caller ID

                                The Caller ID of the Source

                Source Channel

                                The Channel of the Source caller

                Destination          

                                The Nickname the PBX gives the Destination

Destination Caller ID

                                The Caller ID of the Destination

                Destination Channel

                                The Channel of the Destination caller

                DID

                                The DID the Call came in on (if applicable)

               

Ring Group Name

                If it is sent into a ring group, the name of that ring group.

Hold Time

                                The amount of time the party was waiting for an Agent

Status

*Abandoned: Call entered a Queue, but the caller hung up before the call was answered or met the failover requirement.
*ChanUnavail: No channel was able to be created for the call due to a possible error state for the intended destination.
*Agent Completed: Connected Queue call was terminated at the extension.
*Hangup: Call to a Menu, but the caller hung up before taking any actions.
*Caller Completed: Connected Queue call was terminated at the calling parties end.
*Cancel: Call from an Extension, but the caller hangs up before the call is connected.
*Answer: A connected call.
*No Answer: A call that met the ring time of an extension without being answered.
*Timeout Exit: Call to a Queue that meets the Timeout.
*VM Not Left: Call ended with the caller being sent to voicemail where they did not leave a message.
*VM Left: Call ended with the caller being sent to voicemail where they left a message.
*Transferred: Connected call successfully transferred.
*Busy: Call out a trunk returned busy status, or extension was busy.
*- : Defines the call is still active, and the status will update when completed. Click on the call record for details.
*Exited Empty Queue: Call to Queue that follows Exit Empty rule.
*Exited With Key: Call to Queue that follows the Exit Menu key press.
*Account Code: Reserved for future account name information
*Authorization Code: Reserved for future authorization code information

 

         

         

 

 

Example:

The call is incoming on a hardware channel 1 group 1, the call ID links the two records together. The call first goes to a ring group ‘rg_33’ is answered by SIP/2208 (Elaine), and was transferred after 35 seconds of talking. Because the final flag is not set, we know there will be more about this call in the future.

 

The second entry for the call was picked up by SIP/2210 (Chris) and lasted 81 seconds long. The final entry flag has a ‘1’ signifying it is the last leg of this call.

 

1264086258, 1264086296, Incoming, DAHDI/1-1, 1264086247.6308, 7145554445, 7145554445, Zap/1-1, Elaine Blodgett, , SIP/2208, 9413062200, rg_33, 3, TRANSFERRED,35,

 

1264086296, 1264086395, Incoming, DAHDI/1-1, 1264086247.6308, 7145554445, 7145554445, Zap/1-1, Chris Beavers, 2210, SIP/2210-087b12f0, 9413062200, , , ANSWER, 81, 1