VoIP Trunks

VoIP Trunks (also referred to as SIP Trunks) can be used to achieve a significant reduction in outbound toll rates when calling long distance or internationally. They can also be used to link or trunk two systems together so you can call another location as if you were in the same office.

To add a VoIP Trunk:

  1. Select your provider from the "Add Phone Lines" drop down list. If your provider is not listed, select "VoIP-Generic"
  2. Enter your Account ID and Password (not required for IP Only)
  3. Click the "Advanced Options" check-box to show advanced options
  4. Set the "Proxy Address" (also referred to as SIP Proxy or Inbound Proxy)
  5. Select the appropriate "Auth Type"
  6. To the right of the window, under "Inbound Calls" you can enter any DIDs and set a destination for them.
  7. Click "Save"

Account Settings

Setting Description
Account ID The username or ID for credentials based authentication on your VoIP Trunk
Password The account password for credentials based authentication on your VoIP Trunk

G711: The default audio codec. Produces good audio quality at about 160 Kbps of bandwidth per call. Also refereed to as μ-law.
G729: This codec requires a license per channel. It produces good audio quality at a much lower bandwidth of about 48 Kbps per call.
GSM: Produces acceptable audio quality at about 58 Kbps per call.

Note: Bandwidth calculations assume no trans-coding and include both inbound and outbound streams with typical overhead. Kbps stands for Kilobits per second.

Max Calls Select the maximum number of simultaneous calls this VoIP Trunk supports.
Default Provider Check this box to set this VoIP trunk as the default outbound provider on your system.

Advanced Settings

Setting Description
Auth Type Credentials based authentication requires an Account ID and Password that is provided by your VoIP provider. IP Only authentication can be used if your provider supports this method. Do not use IP Only authentication if you have a dynamic IP address.
Proxy Address This is the inbound VoIP/SIP proxy that you will register to and receive calls from.
DTMF Mode DTMF (Dual Tone Multiple Frequency) are the audible tones that you hear when pressing a key. This setting specifies how your provider is expecting to receive these digits. RFC 2833 is a standard for VoIP.
Caller ID Mode Passthrough enables you to set the Caller ID to anything you want when placing an outbound call. If your VoIP provider does not allow you to do this, you will select "Account".
Heartbeat Also refereed to as "Keep Alive", this can help maintain SIP registration and NAT traversal. Select how often in seconds to send the SIP OPTIONS message to the VoIP provider.
OB Proxy

The outbound VoIP/SIP proxy address. Some VoIP providers require you to use a separate proxy for outbound calls.

Outbound Number Translation

The outbound number translations provide a method to change the number as dialed to match requirements met by your provider. You can prepend or remove digits from a dialed number before it is handed to the provider. For instance your provider may require a 1 even if you are doing a 10 digit number. Numbers will show up after the provider has been referenced in the outbound dialmap. Below are a few examples of how to modify the dialed number:

  • Striping the first digit from the beginning of the dialed number: ${DIALED:1}
  • Adding a 1 to the number dialed: 1${DIALED}
  • Adding the area code 949 to the dialed number: 949${DIALED}

Inbound Calls

Here you will enter the DIDs you have with your VoIP Trunk. At first you will only be presented with one line to enter a number; however, as you click on the "Description" of "Number" field, an additional line will show up below. Each DID will be entered with the following options:

Setting Description
Description The description field is used to identify the DID. When calls are received, the caller ID name will be prepended with the description you enter.
Number The 10 digit (US and Canada) DID number without any special characters like spaces or hyphens. You can also use standard Asterisk pattern matching. For example, _X. could be used to route any DID to a particular destination.
Send Calls To Select the type of target that calls will be routed to, when the corresponding number is dialed, from the list of available options. For example, to route inbound calls to an extension group, select "Extension Group".
Location Select the specific resource that you want inbound calls to route to. If you selected "Extension Group" to "Send Calls To", you would select the specific extension group number here.

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