mitel-sip-trunk
mitel-sip-trunk
Configuration Guide
Valcom IP devices can communicate with Mitel MiVoice Business systems over a SIP Trunk
connection. The following example illustrates the programming on a MiVoice virtual 3300 ICP release
9.0 SP3 system and a Valcom VE8090 SIP Controller.
The basic process is to create a Network Element representing the Valcom device, define several
options in the SIP Device Capabilities, define a SIP Trunk Peer Profile and define the Call Routing to
direct calls to the Valcom device. Other settings may be required in the Mitel system for specific site
requirements related to your installation.
Login to the MiVoice Business System Administration Tool and navigate to the Voice Network →
Network Elements page. Fill in the appropriate fields for Name, IP Address and check the box for SIP
Peer. In the Peer Specific area, Transport can be either TCP or UDP. If the default Peer Port of
5060 is changed here, it will also need to be changed in the Valcom VE8090 programming. Save the
entries when completed.
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Mitel MiVoice Business SIP Trunk
Configuration Guide
Navigate to the System Properties → System Feature Settings → SIP Device Capabilities page. If a
SIP Device Capabilities has been defined with the necessary settings, it can be used instead of
defining a new one for the Valcom device. The minimum settings are shown below.
On the Basic page, confirm TLS Only is set for No, and Replace System based with Device based In-
Call Features is set to Yes.
On the SDP Options page, the default values as shown below are sufficient.
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On the Signaling and Header Manipulation page, confirm Disable Reliable Provisional Responses is
set to Yes.
Programming on the remaining Tabs in the SIP Device Capabilities can be left at the default values.
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Navigate to the Trunks → SIP → SIP Peer Profile page, and create a new SIP Peer Profile. Enter a
descriptive name for the SIP Peer Profile Label and select the Network Element created previously
for the Valcom device. Set the Maximum Simultaneous Calls to 4. Other items in the Administration
Options should be reviewed for any site-specific settings, such as Zone. The Authentication Options
and Gateway Options can be left blank.
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On the Call Routing page, set Private SIP Trunk to Yes.
On the Calling Line ID page, set Use Original Calling Party Number if Available to Yes.
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On the Signaling and Header Manipulation page, set Disable Reliable Provisional Responses to Yes,
set Require Reliable Provisional Responses on Outgoing Calls to No.
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Navigate to the Call Routing → Automatic Route Selection (ARS) → ARS Routes page. Select an
available Route Number and click Change. For Routing Medium, select SIP Trunk. For SIP Peer
Profile, select the profile created previously.
Navigate to the Call Routing → Automatic Route Selection (ARS) → ARS Digits Dialed page. Leave
the number of records to add at 1. In the Digits Dialed field, enter the digits that will be dialed by a
phone user to call the VE8090. In this example, 60000 will be used as the phone number to call. For
Termination Type select Route and for Termination Number select the Route Number that was used
in the previous step.
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Mitel MiVoice Business SIP Trunk
Configuration Guide
Programming the Valcom device for SIP communication is done using the VIP-102B IP Solutions
Setup Tool. The information contained in this guide is limited to configuration of the “SIP” tab in the
VIP-102B Setup Tool for the Valcom VIP device. More information on Valcom VIP device
configuration, such as IP address assignment, relay activation, etc., may be found in the VIP-102B
Reference Manual. This document may be downloaded from our website at www.valcom.com
The example below is for programming the VE8090 SIP Controller in Trunk mode.
For SIP Mode select Trunk, and for Transport select the same protocol that was set in the SIP Peer
Transport on the Network Element page. For this example, Accept: TCP+UDP, Originate: TCP would
be the correct choice.
Enter the IP address of the Mitel SIP PBX for the Primary SIP Server. All of the Port values may be
left at default, unless they were changed in the Mitel configuration previously.
The VIP Channel 1 Auto Dest and VIP Channel 2 Auto Dest fields contain the phone numbers of the
Mitel phones that should be called when a Valcom IP speaker places an intercom call to either the
Channel 1 or Channel 2 dial code. In this example, intercom calls from an IP Speaker to the VE8090
Channel 1 will be forwarded to the SIP phone number 4218.
The Secondary Dial Tone check box controls whether the VE8090 will prompt for user input by
playing Dial Tone to the caller, or whether the VE8090 will attempt to route the call based on the SIP
phone number sent to the VE8090. When the Secondary Dial Tone box is checked, the VE8090 will
answer an incoming call from SIP and play dial tone to the caller. The caller will then key in the
Valcom paging group or device dial code they wish to contact. This is typically referred to as “two-
stage dialing”.
If the Secondary Dial Tone box is not checked, the VE8090 will examine the phone number sent to it
in the SIP INVITE message and try to match it to a Valcom paging group or device dial code. With
this method, the Route Pattern defined in the Mitel ARS Dialed Digits page will be defined as one or
more leading digits followed by a set number of additional digits. In our example, instead of the
dialed digits being “60000” followed by no additional digits it could be defined as “60” followed by 3
digits. Calling to 60000 or 60999 or any other 5-digit number starting with 60 would be routed to the
VE8090. The VE8090 will use its Dial Code Length setting (defined under the System menu in the
VIP-102B tool) to determine how many digits of the SIP number to use. The default Dial Code Length
is 3, so for this example a call to 60000 will try to match 000 to a Valcom dial code. A call to 60999
will try to match 999 to a Valcom dial code.
For this example, the Secondary Dial Tone box is ticked and the VE8090 will play dial tone to the
caller.
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