Documentation
pascom Product Documentation and Release Notes
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Desktop Client User Guide
- Log on to the pascom client
- Layout and Overview
- Profile and Contactlist
- Search, Smart Filter and Notifications
- Telephoning with the pascom Client
- Transfer, Pickup and Call forwarding
- Groups for Chat-Rooms and Conferences
- Chat, Group-Chats and Filesharing
- Use Video-Telephony and Screensharing
- Audio- and Video Conferencing
- The Call- and Team History
- pascom Teams - Working with call queues
- The Phonebook
- Send and receive faxes
- Use the Call Recording System
- My Device Control
- Headset and Audio Settings
- Use Kuando Busylight
- How to use Client Actions
- pascom Client Settings
- How to use the tomedo® Integration
- Quick Start
- Server Setup
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Mobile Client User Guide
- pascom Mobile App pairing and login
- Profile and Contactlist
- Search, Smart Filter and Push-Notifications
- Telephoning with the pascom Mobile Client
- Groups for Chat-Rooms and Conferences
- Chat and Group-Chats
- Use Video-Telephony
- Audio- and Video Conferencing
- The Call- and Team History
- pascom Teams - Working with call queues
- The Phonebook
- Mobile App Energy Optimisation Settings
- User Management
- Telephone Book
- Web Conference User Guide
- Team Management
- Client Administration
- Endpoints
- Gateway Configuration
- Trunks
- Trunk Templates
- Automatic Call Distribution
- Modules
- Concepts
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How to
- Data protection and IT operations in the pascom cloud
- Monitoring via Nagios or Icinga
- Setup Mobile Access via the Internet
- Using a Lancom as a Router
- Nextragen - VoIP & Network Monitoring
- Subscription Downgrades
- Setup Extensions
- Remove Dialplan Loops
- Function / Service Codes
- Change SIP Header
- Telephone Provisioning via DHCP
- tomedo® Integration setup
- Support & Release Notes
- REST API
Account Settings
+ Cloud and Self HostedIn the account settings, enter the registration details with which pascom should register on the provider’s server. It is also possible to configure additional settings aside from the authentication details.
The Account Settings
You can access your trunk account data via > > . Under the Account tab you will find the records of your registrar. You can change these as you wish.
Extension Register
| Variables | Description |
|---|---|
| Extension reg. | This option shares the extension which should be called by the host when receiving an incoming call. This is required if your User SIP Account only provides one number. If you use a SIP Trunk connection (e.g. sipgate trunking) then do not enter anything here as the called extension will be transmitted separately. |
If you use a SIP Trunk that contains multiple individual numbers, you will need to create separate data sets for each number under so that each number can register with the provider.
Username
| Variables | Description |
|---|---|
| Username | Depending on your trunk type, the username is used to either * register your phone system with the host (Type: User) or is the name with which a host registers with your phone system (Type:Peer). If type Friend is used, two objects will be created and the username will be used for both the User and the Peer. |
Password
| Variables | Description |
|---|---|
| Password | Is the password for the corresponding user - if required. |
Host
| Variables | Description |
|---|---|
| Host | Enter the IP address or the host name on which the account must register. |
Options
| Variables | Description |
|---|---|
| Options | Options provide you with the ability to add further SIP options (one per line). These options must be compatible with Asterisk phone systems. |
An overview of SIP options can be found here: SIP Configuration - peers and clients
Registration
| Variables | Description |
|---|---|
| Registation | Select Yes or No, depending on whether or not your phone system must register with the host. |
Port
| Variables | Description |
|---|---|
| Port | Port 5060 is used per defualt. Should a different port be used, you can enter the port here. |
Number Determination
| Variables | Description |
|---|---|
| Number Determination | The default option to determine the destination number using the SIP-INVITE. Set this option to yes and the destination number will be read from the SIP-Header instead. This option is often required by SIP trunk providers. |
Number Presentation
| Variables | Description |
|---|---|
| Number Presentation | Calling Line Identification (CLIP) is a feature that enables the transmission of phone numbers for outgoing calls. By default, the automatic Trunk functions take effect, so that the trunk can automatically transmit call numbers. Possible settings: * AUTO * FROM * PPI * PAI |
Number Restriction
| Variables | Description |
|---|---|
| Number Restriction | Calling Line Identification Restriction (CLIR) is a feature that enables call number restriction for outgoing calls. The automatic Trunk function comes into play here as standard, so that the trunk can automatically restrict the number. Possible settings: * AUTO * FROM * Privacy-ID |