Difference between revisions of "SIP Trunk (Zenitel Connect Pro)"
From Zenitel Wiki
(Created page with "{{C}} ==Introduction== Zenitel Connect Pro (ZCP) can be interconnected to a SIP Server (e.g. iPBX or Call Manager) via a SIP trunk. There are two ways to call from a user in...") |
Lennert.werf (talk | contribs) (→Adding the SIP Trunk and defining the Prefix) |
||
Line 15: | Line 15: | ||
* '''Prefix''': The digit(s) to dial to access the trunk | * '''Prefix''': The digit(s) to dial to access the trunk | ||
* '''Name''': Any descriptive text | * '''Name''': Any descriptive text | ||
− | * '''Peer Address''': The IP address of the SIP Server | + | * '''Peer Address''': The IP address of the SIP Server. If the peer uses a port other than 5060 or 5061 it can be added to the address. |
* '''Call Service''': [[Permissions and Call Services (Zenitel Connect Pro)|Call Service]] for an incoming call. | * '''Call Service''': [[Permissions and Call Services (Zenitel Connect Pro)|Call Service]] for an incoming call. | ||
* '''SIP Transport''': protocol to use (UDP, TCP or TLS) | * '''SIP Transport''': protocol to use (UDP, TCP or TLS) |
Revision as of 11:10, 30 October 2024
Introduction
Zenitel Connect Pro (ZCP) can be interconnected to a SIP Server (e.g. iPBX or Call Manager) via a SIP trunk.
There are two ways to call from a user in ZCP out on the SIP trunk:
- Prefix: Dial the Prefix + the external extension number
- Integrated number plan: Simply dial the extension number without any prefix
SIP Trunking is a licensed feature. A ZCL-PBX license is required per trunk |
Adding the SIP Trunk and defining the Prefix
Go to Devices and Connections > External Communication, select Trunk Type External iPBX communication and click the to add a new trunk.
- Prefix: The digit(s) to dial to access the trunk
- Name: Any descriptive text
- Peer Address: The IP address of the SIP Server. If the peer uses a port other than 5060 or 5061 it can be added to the address.
- Call Service: Call Service for an incoming call.
- SIP Transport: protocol to use (UDP, TCP or TLS)
- SIP Authentication: Username and Password to authenticate towards the SIP server
- Medias codecs: Options are G722, G711u, G711a, GSM, H264, L16x48
- Media encryption: Choose No or SRTP (secure RTP)
- Status: Reachable or Unreachable. Tells if the connection is up or down.
Defining a SIP trunk |
Integrated Number Plan
It is possible to define outgoing number ranges. Numbers in these ranges can be dialed directly, without the need to dial a prefix. This gives a possibility to make a integrated numbering plan for all SIP trunks in a linked network.
To define the outgoing numbers, select a defined trunk and click on on the left.
In the Outgoing Numbers Range pane, click on to define a new range and enter the From and To directory numbers.
Here the numbers 230-299 will be routed to the SIP Trunk |