The following Configuration Guides are intended to help you connect your IP Communications Infrastructure (Contact Center, IP-PBX, SBC, etc) to a Lineblocs hosted SIP Trunk.
We have configuration guides with the following types of IP communications infrastructure elements:
Assuming you have Asterisk already set up as your IP-PBX, with one or more telephones configured and running calls between them, the following guide provides detailed step-by-step instructions on how to configure your Trunk and your Asterisk IP-PBX.
Click here to download the Asterisk Interconnection Guide
Assuming you have FreeSwitch already set up as your IP-PBX, with one or more telephones configured and running calls between them, the following Interconnection Guide provides you with step-by-step instructions to use FreeSwitch PBX with your Lineblocs SIP Trunk.
Click here to download the FreeSwitch PBX Interconnection Guide
Assuming you have your 3CX already set up with one or more telephones configured and running calls between them, the following highlights specific configurations for use with your SIP Trunk.
Add a new VoIP Provider account in the 3CX phone system:
Set the SIP server hostname to:
Set your Authentication ID/username and password (as you configured in your user credentials)
DID's and Inbound Call Identification: Enter your numbers under the "DID" tab.
"Advanced" under "Codec priorities" only include G711 U-law
Create Outbound Call Rules: setting calls to numbers with a length of 10, and also prepend a "+1". This will ensure E164 formatting.
Click here to download the 3CX Interconnection Guide
Assuming you have FreePBX already set up as your IP-PBX, with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your Trunk.
Click here to download the FreePBX Interconnection Guide
The following Interconnection Guide provides you with step-by-step instructions to use GrandStream UCM with your Lineblocs SIP Trunk.
Click here to download the Grandstream Interconnection Guide
The following Interconnection Guide provides you with step-by-step instructions to use Genesys Cloud BYOC your with Lineblocs SIP Trunk.
Click here to download the Genesys Cloud Interconnection Guide
Zoom Phone is an enterprise cloud phone system. Zoom Phone offers two "bring your own carrier" (BYOC) approaches, called "Zoom Phone Premise Peering PSTN" and Zoom Phone Carrier Peering PSTN, which both provide organizations the flexibility to select their voice services for Zoom Phone.
Click here to download the Zoom Phone Configuration Guide
The following guide is not maintained by Lineblocs. Please see Mitel Knowledge base for the latest guide.
Click here to download the Mitel MiVoice Configuration Guide
Navigate to "VoIP">"SIP" to configure the SIP server info for Lineblocs. Enter in the SIP Server FQDN assigned for these services under the SIP Server Address field. Fill in the SIP Server Domain field with the proper Lineblocs domain.
Note: Make sure to check the "Limit Inbound to listed Proxies" and "Limit Outbound to listed Proxies" boxes to help prevent fraudulent activity sourced from a LAN side PBX or a WAN side DoS attack.
Navigate to "VoIP ALG" and then "B2BUA" to configure the SIP Trunk registration with the soft-switch (between the Ribbon EdgeMarc and the WAN side soft-switch), the PBX for SIP registration mode (between the PBX and LAN side of the Ribbon EdgeMarc), inbound rule (for sending SIP messages from the WAN side of the Ribbon EdgeMarc to the PBX) and outbound rule (for sending the SIP messages from the Ribbon EdgeMarc to the WAN soft-switch). RFC-4904 support will be handled by applying header manipulation action rules to the matched outbound rules.
Configuring the PBX for SIP registration mode (between PBX and the Ribbon EdgeMarc). From the "Trunking Devices" section:
Configure the Ribbon EdgeMarc default inbound rule (for sending the SIP messages from the Ribbon EdgeMarc to the PBX). This is required in order for non-pilot DIDs to reach the PBX.
From the Actions section:
From the Match section:
From the Match section:
Assuming you have your SBC already set up with your IP-PBX, with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your Lineblocs trunk.
Make sure you have your Network & Physical Interfaces appropriately configured.
Configure your Trunk SIP Interface towards Lineblocs:
Configure your Session Agent towards Lineblocs:
The second example presented here illustrates adding +1 to called numbers (To and Request-URI headers) for all SIP trunk endpoints in a particular realm.
Firstly, define the session translation with a called rule:
Then define the rule to append +1:
Lastly, apply the translation as outgoing to the SIP trunk realm:
Set the preferred codec to G711 mu-law. In the example below, the Net-Net SD manipulates the codec list for all PBXs in the PBXs realm such that PCMU appears first in the media descriptor offered to the SIP trunk:
Cisco ISR (Cisco 28xx, 29xx, 38xx, 39xx, 43xx etc.)
Assuming you have your ISR already set up with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your LineblocsTrunk.
Update your Trust List:
Assuming you have your SBC already set up with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your LineblocsTrunk.