PBX-specific configuration: Setting up softphone functions (SIP) for Unify OpenScape Business – connection instructions
This guide describes how to configure softphone features (SIP) for ProCall Enterprise in conjunction with an OpenScape Business from Unify.
ProCall Enterprise | from V 6.2 |
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PBX Unify OpenScape Business | V2 R4.0.1 |
Notes on the estos test environment
The softphone functions (SIP) of ProCall were tested in the estos test environment with the telephone system specified above. The tests were performed with a maximum of two lines per SIP end device. The following information was used during the setup of the telephone system for the login/registration of the individual SIP lines to the UCServer.
Note on codecs and UDP connections
For the connection of SIP lines to the UCServer, the telephone system must support the G.711 a-law codec for the SIP protocol and should have this set as the preferred codec for all end devices.
Only UDP connections are supported.
As of ProCall Enterprise 7.5: Support of Codec Opus and G.722: HD Voice/HD Telephony with ProCall Enterprise SIP softphone functionality
Note on the use of the G.722 codec on the OpenScape Business
The G.722 codec was set up by changing the codec priority on the OpenStage phone. There was no configuration on the OpenScape Business PBX.
Note on the use of call forwarding in the UCServer
If the Call forwarding feature is used in the UCServer, the SIP lines must allow second calls in order to use the Call forwarding on busy feature.
Configuration of the telephone system OpenScape Business
The lines to be used for ProCall must be set up as participants with type SIP Client under menu item Expert mode – Telephony.
Each of the SIP lines/subscribers used for ProCall requires the activation of a license for IP subscribers, which can be activated via the license management.
To set up the SIP line in the UCServer, you need the following information/settings under End device/participant setup – IP end devices:
Participant's phone number | Menu: Setup – Wizards – End devices/Participants – IP Terminals |
SIP User ID/Username It is recommended to enter the phone number. | Menu: Security |
Optionally, authentication can be activated and a password can be assigned. |
UCServer configuration
The settings are made in the UCServer administration under Telephony – Lines:
- Add telephone system
Select Unify OpenScape Business. - Enter the connection to the PBX under Softphone registrations as "SIP Softphone“.
All SIP lines must be added in this connection setting. For the connection to the Unify OpenScape Business, the following must be specified:
If SIP User ID equals phone number as Username the "Phone number" as Password the "Password"
Optional specificationIf SIP User ID is not equal to the phone number as Username the "Phone number" as Password the "Password" as Auth. Username the "SIP User ID"
From ProCall 6.1, please note the following settings
On the SIP connection line group it is possible to define what the UCServer signals to the PBX when:
- The client is not logged in or call protection is enabled
- The call is rejected by the client or no devices are available
The required settings depend on the telephone system and its configuration.
Example: The telephone system should redirect incoming calls to the mailbox if the ProCall client is not logged in.
Problem: The PBX does not evaluate the busy here (486) sent by default from the UCServer and the calls are not redirected.
Possible solution: In the PBX, this setting is set up for Participant unavailable, and Temporarily unavailable (480) is selected for signaling on the UCServer.
Telephone system restriction
Please note that the PBX does not support multiple SIP user agents registered to one SIP extension.
As an example: the UCServer with IP address 192.168.1.1 and port 50000 is registered on extensions 100-200 of the PBX. Now any other SIP end device is registered to extension 100. This will cause all registrations of IP address 192.168.1.1 with port 50000 to be cancelled. In this case, the UCServer would lose control of extensions 100-200. If any calls are made, they would be terminated.
Depending on the reregister time set and negotiated, the problem can keep building up because the SIP end device described above and the UCServer keep reregistering and deleting each other's registrations.
Version note
Since estos has no influence on the further development of the supported telephone systems by the manufacturer, we cannot guarantee that the instructions described above will also be fully valid for future releases.