Please enable JavaScript to view this site.

IP Office Basic Edition - Web Manager R11.1

> Configuration Menus > System > SIP Trunks

Call by Call Settings

Scroll Prev Up Next More

This menu is accessed by selecting System and then SIP Trunks from the menu bar. Select the trunk required and click on Details. Click on the web edit icon edit icon in the Call by Call panel.

web sip call by call

SIP Call By Call List

These settings are used to match calls received on SIP trunks channels set to Incoming Call-by-Call. For systems operating in Key System mode, the default entry is used for all calls for which there is no other match and is fixed to route those calls to the Operator Group.

ARS
This setting is only shown for PBX mode systems. For those systems, each call-by-call entry can be assigned to an ARS Selector number. That selector number can then be used as the destination for outgoing calls.

Local URI:
The user part of the SIP URI. This specifies the contents of the FROM field when making a call (sending an INVITE).

Destination
Where incoming calls with matching digits should be routed. The drop-down list contains the extensions and groups on the IP Office system.


Extension
Route incoming calls to a particular extension.


Phantom Extension
A phantom extension can be selected as the destination for calls.


Calling Group
For systems with their Mode set to PBX, incoming calls can be routed to one of these 4 ring all groups.


Operator Group
For systems with their Mode set to PBX, incoming calls are routed to the Operator Group.


Voicemail
Route incoming calls to the systems voicemail to collect messages. This requires the caller to know the mailbox number and passcode.


76: Modem
The option 76: Modem can be selected to route the call to the systems built in V32 modem function. This is only intended for basic access by system maintainers.


Auto Attendant
Any configured voicemail auto attendants can be selected as the call destination.

Authentication Name
When making a call, some service providers will often send an authentication challenge to validate the call before it is connected. This challenge requires the INVITE is re-submitted with Authentication data, including a network account name (provided by the service provider during installation). The network account name is the “Auth name”. It can be blank, in which case the Local URI is used.

Password: Default = Blank.
This value is provided by the SIP ITSP.

Details
This control can be used to display additional settings associated with the call by call entry.
web sip call by call details

Display Name: Default = Use Authentication Name
This field sets the 'Name' value for SIP calls using this URI.

P-Assert-ID
If this field is configured, the channel can be used in SIPConnect Option 1 model for separating Public and Private PSTN identity (Sipconnect technical recommendation v 10, section 12.1.1). You can only use Explicit CLI configurations over SIP if using Option1 model for identity. In this case, calls over this channel will always have a fixed P-Assert-ID, but the From field may vary.

Registration Required
When selected, each local URI with unique Authentication credentials will register independently.

 

 

Performance figures, data and operation quoted in this document are typical and must be specifically confirmed in writing by Avaya before they become applicable to any particular order or contract. The company reserves the right to make alterations or amendments at its own discretion. The publication of information in this document does not imply freedom from patent or any other protective rights of Avaya or others. All trademarks identified by ™, ® or © are registered trademarks or trademarks respectively are the property of their respective owners.

© 2020 AVAYA
Issue 05.a.–
Thursday, July 23, 2020