IP PBX Manual Providers

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Providers

Providers provide telephone network connectivity, (a.k.a PSTN connectivity).  Examples of providers:  an analog telephone line, or a SIP Provider or a T-1 PRI.  VoIP providers and other telecommunication resources. This section of the system is where these provider resources are configured. The PBX system is equipped to handle two types of providers

1. Hardware
2. SIP Providers

Hardware Providers

Hardware providers or trunks are associated with telephone lines that connect to the PSTN. These lines process inbound and outbound communication traffic that flows over communication channels. For example, a T1 can be a trunk resource that has multiple lines and multiple Direct Inward Dialed (DID) numbers. These individual numbers can be routed to different destinations within the system.

Connection Types

The IP PBX is equipped to support an assortment of hardware cards as discussed earlier. Depending on which card you have, the PBX can integrate with analog lines, T1, or PRI.

Provisioning a New Hardware Trunk Group

Provisioning a hardware trunk defines how the system works with the provider equipment, tells the system what phone numbers are associated with the trunk, and establishes rules for the system to follow when processing incoming and outgoing calls through this physical network connection.

The following table describes the fields and functions available on the Hardware Trunks Provider Configuration page.

Section	Description
Configuration Section
Slot	Defines which physical PCI slot the card is connected to.
Channels	This indicates whether the card has channels configured. Using the mouse to hover over this field will provide you with the number of channels at are set for this card.
Model	This is the model or type of card.
	This button changes the card set in the database for the adjacent slot. The change is immediate and will delete trunks associated with another card in the same slot.
	Selecting this button will take you to the Channel Group Parameters Card Configuration page, where you can create a channel group.
Channel Groups Section
Name	The name associated with this channel group.
Channels	The number of channels that have been assigned.
Card	The card number associated with the channel group.
Action	- Clicking on the pencil icon allows you edit the settings for the selected channel group. - Clicking on the “X” icon allows you to delete/remove the selected channel group. NOTE: You must delete channels from outbound routes in order to delete them.
T-1 Span Configuration
Span	Should automatically set to the appropriate span in regards to the cards that are installed in the PBX.
Timing	Set per your providers instructions, “0” (zero) would be the PBX provides timing, “1” would be the timing is provided by the Provider.
LBO	Line Build Out, set per your providers instructions.
Framing	Set per your provider’s instructions.
Coding	Set per your provider’s instructions.
PRI	Check if the trunk is a PRI, leave blank if the trunk is a standard T-1.
T-1 PRI Configuration
Span	Should automatically set to the appropriate span in regards to the cards that are installed in the PBX.
Switch Type	Defines what type of equipment the PBX is communicating with. Set per your provider’s instructions.
Reset Interval	Set per your provider’s instructions.
Dialplan	Set per your provider’s instructions.
Indication	Set per your provider’s instructions.
D Channel	Set to 24.
USB Device Browser
Slot	Should automatically set to the appropriate slot that corresponds to the USB slot the device is connected to.
Hardware Functions
Restart USB Devices & PBX Services	This button will restart USB devices and PBX Services.

Table 12Hardware Trunk Provider Settings and Descriptions

Configuring Hardware Trunks

STEPS:

1. Navigate to ProvidersHardware Trunks. The Hardware Trunk (Providers) page appears. The hardware trunks and their values that have already been provisioned will be displayed.
2. The system will allow configuring of up to 2 cards that interface with analog or T-1 devices. Select the card models installed in your IP PBX individually and click Set button for each card in order to tell the system which cards you have.
3. Once you have set the card, click the Add Lines button to the right of the corresponding card to define line/channel/trunk groups associated with that card. The Edit Hardware Provider page appears.

1. Edit the necessary parameters. Required fields are: Group Name, Signalling Type, Start Channel, and End Channel. The rest of the fields can be configured as needed, or left at their defaults. (refer to the table below for descriptions and recommended settings)
2. Click on the button.
3. Click the Apply Changes link located on the right hand corner of the page to commit the changes to the database.
4. Once the changes are applied, navigate to the Hardware Providers page and click the Restart USB and PBX Services button.

The following table describes the fields and parameters available on the Add/Edit Hardware Provider Configuration page:

Section/Field	Description
Channel Group Parameters Card 1
Group Name	This is the name for the groups of channels you want to create. This Group Name will be associated with the line/channel group.
Signaling Type	This is the type of signaling used for these channels/lines. Recommendation and default settings are listed below: Analog lines should be set to fxs_ks. T-1 lines should be set to fxs_ls or em_w T-1 PRI lines should be set to pri_cpe.
Generate Ringing on Outbound Calls?	This should only be checked if the telephone company is not providing ringing. Default this is disabled.
Answer Incoming?	This indicates whether the system should answer incoming calls on these lines/channels. The default value is Yes.
Start & End Channel	Used to define the Start and End channel for this group on the associated card. If you would like a 1 channel group set the start and end channel to the same value (i.e. Start=1 and End=1). IMPORTANT: If you wish to address analog lines individually you must build multiple single channel groups. T1 and PRI can be grouped with multiple channels and routed via DID.
Allow Caller to transfer outbound call?	This indicates whether the system will allow user to transfer a call they origninated that has been connected to the PSTN. Default this is disabled.
Answer After	If the system is configured to answer incoming calls, this defines how many seconds to wait before answering the call. At times this field needs to be tweaked to ensure consistent CID. Default this is set to 0.
Dial Prefix	Digits defined here will be dialed out the trunk ahead of the digits dialed by the user. This would be required if your provider needs a 9 or some other digit to dial outbound. Analog trunks might need a one second pause which is denoted by a lowercase w. Default this is blank.
Allow Call Recording?	This setting toggles the ability to record calls on this trunk. Default this is disabled.
Ext CID Override?	This setting allows the users to set an alternate caller ID under their extension that will override the outgoing caller ID. If this is enabled, then this allows extension caller id override capabilities. Default this is disabled. IMPORTANT: Contact your provider to determine whether CID override is allowed on your specific trunk
Restrict CID Override?	This setting is used to limit an extension’s ability to override Caller ID (CID). If enabled, this allows extension CID Override only for the phone numbers defined for this provider. Default this is disabled.
Use Caller ID	This setting indicates whether the system should attempt to detect caller id on the lines/channels in this group. Default this is set to Yes.
Echo Cancellation	This allows you to configure the Echo Cancel parameter. By default this is set to Yes (128) which will work in most scenarios. Adjustments can be made as needed.
RX Gain	This allows you to set or adjust the gain on sound received from these lines/channels. Typically these settings don’t need to be changed from default settings, unless there are issues relating to the volume on the calls. (This setting is in decibels so 3.0 = 100% increases in volume). Default this is set to 0.0.
TX Gain	This allows you to set or adjust the gain on sound transmitted to these lines/channels. Typically these settings don’t need to be changed from default settings, unless there are issues with the volume on the calls. (This setting is in decibles so 3.0 = 100% increases in volume). Default this is set to 0.0.
Inbound Caller ID	This is the name of the Caller ID of the inbound calls on this trunk. If set to the system default “asreceived', the calls will capture the Caller ID information sent over the incoming lines. Otherwise you can override the name and number that appears to those receiving calls on the corresponding channels. IMPORTANT: If you need to change this parameter, we recommend that you contact IPitomy’s Technical Support Group at (941) 306-2200 for the proper syntax or additional information.
Echo Cancellation Bridged	This allows you to enable or disable the Echo Cancellation Bridge parameter on calls routed internally across the network. Set this parameter to”YES” (enable) if you want echo cancellation on calls that are bridged from one port to another on the network. Default this is set to No.
Busy Detect	By default, this is set to YES and8. This allows you to enable/disable and set the number of busy signals the system will detect before disconnecting from the call (hanging up). Set the parameter to “YES” if you want the PBX to hang up after it detects a busy signal. The number field is the number of busy tones the system must detect before it disconnects from the call (hangs up).
Outbound Caller ID Name	Any values entered here will display as the Outbound CID Name when making calls out this trunk. If this is left blank, the name displayed on outbound calls will match what is registered with the provider. Blank by default. IMPORTANT: Contact your provider to determine whether CID override is allowed on your specific trunk
Outbound Caller ID Number	Any numbers entered here will display as the Outbound CID number when making calls out this trunk. If this is left blank, the number displayed on outbound calls will match what is registered with the provider. Blank by default. IMPORTANT: Contact your provider to determine whether CID override is allowed on your specific trunk
Echo Training	This parameter is used to adjust the time spent training the echo cancellation algorithm before a call is picked up. The length of time is set in milliseconds (ms), and is set to 800 by default.
Route Calls to	This is the default destination for the trunk. A destination is where calls from this provider will be routed to, unless a specific Direct Inward Dial (DID) is indicated. A typical destination is defined as a Menu, Extension, Schedule, Voicemail, or a Ring Group. This can also be set under the Incoming Routing page.
Relax DTMF (Detection)	This allows you to turn On/Off the Relax DTMF detection. Enabling this setting allows the line/channels to be more permissive of tone lengths and result in over detecting. By default this is set to No. IMPORTANT: We recommend that this parameter is set to NO unless you are having issues. Please contact IPitomy’s Technical Support Group for assistance or more information.
DTMF Tone Duration	This allows you to set the DTMF tone duration in milliseconds (ms) for generated tones generated by calls bridged to SIP devices. Default value of 300.
Use Inbound CoS	This allows you to enable/disable inbound routing and provides the dial plan that corresponds to the outbound routes in this Class of Service (CoS). This is Off by default. IMPORTANT: Use this only if you intend to provide PSTN connectivity to 3rd party equipment. Please contact IPitomy’s Technical Support Group for assistance or more information.
Phone Numbers	On a T1 or PRI, this is where you would configure any DID information. Only numbers or a + should be entered for a DID. Once configured, you can define a different inbound destination for each number, allowing you to route calls to separate places. Contact your provider to find out if they are sending 4, 7, or 10 digits inbound for the DID.

Add Phone Numbers to Hardware Provider

Once all the destinations are created, they will appear in a drop-down list and can be selected for updating.

STEPS:

If a DID destination is set to NONE, it will route to the default location for the trunk itself.

1. From Channel Groups section of the Hardware Provider page, click the icon to the right of the name you want to modify. The Edit Hardware Providers page appears.
2. The hardware trunks and their values that have already been provisioned will be displayed. Scroll down the page to the Phone Numbers section. Enter the phone number that you want to add then click the ADD button. You can also copy and paste a list in. The list should be separated by newline characters.
3. The phone number entered will appear in the drop-down list to the right of the ADD button. The numbers will be listed in sequential order.
4. Click on the button to save the changes to the system.
5. Click the Apply Changes link located on the right hand corner of the page, to commit the changes to the database.

Remove Phone Numbers from Hardware Provider

STEPS:

1. From Channel Groups section of the Hardware Provider page, click the icon to the right of the name you want to modify. The Edit Hardware Providers page appears.
2. Scroll down the page and click on the field to the left of the REMOVE button. Scroll through the drop-down list to find the phone number that you want to remove.
3. Click on the desired phone number. Click on the REMOVE button to the right of the list. The phone number is deleted and will not appear in the drop-down list.

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

1. Click on the button to save the changes to the system.
2. Click the Apply Changes link located on the right hand corner of the page, to commit the changes to the database.

Provisioning a SIP Provider

SIP Providers are VoIP service provider accounts or other SIP-based devices that provide PSTN connectivity. SIP provider accounts can have multiple phone numbers or Direct Inward Dialing numbers (DIDs). The individual numbers can be routed to different destinations within the system as DIDs.

IMPORTANT: SIP Providers settings vary widely. To simplify the provisioning process, we recommend that you ask the provider for a sample Asterisk configuration. Please contact IPitomy’s Technical Support Group for assistance.

The following table describes the fields and functions available on the SIP Provider Configuration page:

Sections/Fields	Description
SIP Provider Section
Name	Name assigned to the SIP Provider. If given a username, this needs to match that field.
User Type	User type of the associated SIP Provider. This can be User, Friend, or Peer.
DTMF Mode	DTMF tone for the trunk. This can be Inband, rfc2833, info, or auto.
RFC2833 Compensate	This advanced feature is sometimes necessary when using RFC2833 is problematic or if the SIP provider has indicated that it is required.
Host	The domain or IP address associated with the SIP host.
Port	If your SIP provider requires registration on a port other than the default 5060, set this field to Custom and enter that port value; otherwise set to Default. NOTE: Be sure to forward this alternate port in your router or the traffic will not be passed through.
Register	Registration option for the SIP Provider. YES = Automatically generates based upon provided settings NO = Doesn’t require authentication from the SIP Provider CUSTOM =Allows for any special authentication rules required from the SIP Provider
Authentication	Authentication (previously “Authorization”) requirements for the SIP Provider. YES = Automatically generates based upon provided settings NO = Doesn’t require authentication from the SIP Provider CUSTOM =Allows for any special authentication rules required from the SIP Provider
Authorized User	Some providers require a different Authorization User Name than the Username provided. The sample asterisk configuration from your provider will let you know if this field needs to be set to Custom; otherwise set to Default.
From User	The sample asterisk configuration from your provider will let you know if this field needs to be set to Custom; otherwise set as Default.
From Domain	This parameter is rarely used and if needed, the SIP Domain information can be obtained from your SIP Provider
Realm	If your provider requires you to set the Realm to something other than asterisk, set to Custom and enter the realm given; otherwise this can be set to Default.
Outbound Proxy	Enable if the SIP Provider requires you to provide a different outbound IP address, otherwise leave as Default.
Username	Username used for Registration and Authentication.
Secret	The password used for SIP Registration and Authentication.
Inbound Caller ID	This parameter should be left blank to display incoming caller ID as received. Otherwise you can override the name and number that appears to those receiving calls on the corresponding channels. IMPORTANT: If you need to configure the system in a different way, please contact IPitomy’s Technical Support Group for assistance and the proper syntax needed. Contact us via phone at 941-306-2200 option 2 or via email at support@ipitomy.com. Additional information can be found at faq.ipitomy.com.
Outbound Caller ID Name	Enter an outbound Caller ID name that will override the Caller ID name that is displayed on outgoing calls through this provider. IMPORTANT: Contact your provider to determine whether CID override is allowed on your specific trunk
Outbound Caller ID Number	Enter the Caller ID number that will override on outgoing calls through this provider.. IMPORTANT: Contact your provider to determine whether CID override is allowed on your specific trunk
Call Limit	Set to match the number of concurrent calls allowed by the SIP trunk. (Usually consistent with SIP provider subscription limit.)
Qualify	This is the number of milliseconds (ms) the system should wait before checking to see if the SIP provider is available. Check with your provider to see what value this should be set to. Typically this can be left at the default value of 30000.
Default Destination	This is where all inbound calls will be routed to, unless a different specific destination is designated for DIDs on an individual basis.
Dial Prefix	Set this parameter only if the SIP Provider requires it.
Area Code	This parameter is obsolete and will be removed in future releases.
Generate Ringing On Outbound Calls	This parameter should only be enabled (checked) if ringing is not provided by the SIP Provider.
Allow Outbound Caller to transfer	Enabling this parameter allows a user to transfer a call that they originated on this trunk. Default is disabled.
Allow Call Recording	If enabled (checked), this parameter allows calls to be recorded.
Ext CID Override	If enabled (checked), this parameter allow the users to set an alternate caller ID under their extension that will override the outgoing Caller ID. Default is disabled. IMPORTANT: Contact your provider to determine whether CID override is allowed on your specific trunk
Restrict CID Override?	This setting is used to limit an extension’s ability to override Caller ID (CID). If enabled, this allows extension CID Override only for the phone numbers defined for this provider. Default this is disabled.
Can Reinvite	This parameter allows a mechanism to reconnecting calls midstream. YES = if the phone type allows the re-invite feature NO = if the phone type does not allow the re-invite feature N/A = accepts the system wide default defined under PBX SetupSIPAdvanced. If the default setting is acceptable and works within your business, we recommend leaving the parameter set to N/A.
Send Remote Party ID	Usually not required. Leave on “N/A” unless advised differently by tech support.
Trust Remote Party ID	Usually not required. Leave on “N/A” unless advised differently by tech support.
Insecure	This parameter allows you to specify how to handle connections with peers. Explanation of the different options available on the drop-down list are: PORT = Ignore the port number where authentication came from. INVITE = Do not require the initial invite to authenticate. PORT+INVITE = Do not require initial invite to authenticate and ignore the port where the request came from. YESTo match a peer based by IP Address only and not the port. VERYTo allow registered hosts to call without re-authenticating.
Allow Codecs	Encodes a stream or signal for transmission. Select which codecs will be enabled for this provider. Can typically be left to defaults.
Phone Numbers Section
Phone Numbers	This is where you would configure any DID information. Only numbers or a + should be entered for a DID. Once configured, you can define a different inbound destination for each number, allowing you to route calls to separate places. Contact your provider to find out if they are sending 4, 7, or 10 digits inbound for the DID.

Add New SIP Provider

The following section outlines the steps to add a new SIP Provider in the PBX system.

STEPS:

1. Click Providers=>SIP Providers. The SIP Providers page appears.
2. Click on the button. The Edit SIP Provider page is displayed.
3. Configure the pertinent fields per your SIP Providers instructions. See the table above for descriptions.
4. Click on the button.
5. Click the Apply Changes link located on the right hand corner of the page, to commit the changes to the database.

Add Phone Numbers

The following section outlines the steps to Add phone numbers to the SIP Provider.

STEPS:

1. Click Providers=>SIP Providers. The SIP Providers page appears.
2. From the SIP Providers page, select the SIP provider link (under the Name field) or the icon to the left of the provider you want to modify. The Edit SIP Providers page appears.
3. Scroll down to the Phone Numbers section of the page. Enter the phone number in the box above the ADD button then click ADD. The number enter will appear in the list to the right of the box.
4. To add multiple phone numbers at one time, press the ENTER key to move to the next space on the list then enter the number. Click on the ADD button and all the numbers will appear in the list box (on the right). You can also copy and paste a list in. The list should be separated by newline characters.
5. Click on the button, once all the numbers have been added, to save the changes.
6. Click the Apply Changes link located on the right hand corner of the page, to commit the changes to the database.

Remove Phone Numbers

The following section outlines the steps to remove phone numbers from the SIP Provider.

STEPS:

1. Click Providers=>SIP Providers. The SIP Providers page appears.
2. From the SIP Providers page, select the SIP provider link (under the Name field) or the icon to the left of the provider you want to modify. The Edit SIP Providers page appears.
3. Scroll down to the Phone Numbers section of the page. Select the number that you want to remove from the list then click the REMOVE button. The number enter will be deleted from the list.
4. To remove multiple phone numbers, hold down the SHIFT key to select the numbers in sequence or the CTRL key to select in random order. Click the REMOVE button to delete the numbers selected.
5. Click on the button, to save the changes.
6. Click the Apply Changes link located on the right hand corner of the page, to commit the changes to the database.

Set Destination

The following section outlines the steps to set a Destination for DID phone numbers.

STEPS:

1. Click Providers=>SIP Providers. The SIP Providers page appears.
2. From the SIP Providers page, select the SIP provider link (under the Name field) or the icon to the left of the provider you want to modify. The Edit SIP Providers page appears.
3. Scroll down to the Phone Numbers section of the page. Select the number that you want to assign a destination. Select the desired Destination from the drop-down list then click the SET button.
4. To assign the same destination to multiple phone numbers, hold down the SHIFT key to select the numbers in sequence or the CTRL key to select in random order. Click the desired Destination from the drop-down list the click SET button. You can also set the destinations under Call Routing=>Outgoing.
5. Click on the button, to save the changes.
6. Click the Apply Changes link located on the right hand corner of the page, to commit the changes to the database.
   1. === Delete SIP Provider ===

The following section outlines the steps to DELETE a SIP Provider in the PBX system.

IMPORTANT: You must remove all instances of the SIP provider from any outbound dialing routes before you can delete it.

STEPS:

1. Click Providers=>SIP Providers. The SIP Providers page appears.
2. From the SIP Providers page, icon to the left of the provider you want to delete. The SIP provider will be removed and will no longer appear on the list.
3. Click on the button, to save the changes.
4. Click the Apply Changes link located on the right hand corner of the page, to commit the changes to the database.