Under [Options > Phone Dialog (TAPI Device) > Select > VoIP] select the "SIP SoftPhone".
- The entry is only visible / available in the Professional version!
- A new SIP registration is to configure via the button [Device Configuration].
- Later you reach the configuration via: [Options > PhoneDialog (TAPI Device) > Driver Configuration > Edit].
SIP Registrations / Accounts / Identities
On the first configuration level you can create one or more SIP registrations / accounts.
User ID / authentication name / password
The necessary data for "User ID, Authentication Name and Password" can be obtained from your SIP service provider.
The user ID often uses the phone number, but it can also be a customer ID, for example.
Specifying an authentication name is optional. If no authentication name is entered, the user ID is used for the authentication process.
Several SIP Accounts
If you have configured multiple SIP accounts:
If you have configured several SIP accounts, the "Phone number(s)" button should be activated under [Options > PhoneDialog (TAPI device)]. Check the boxes here for the phone numbers/accounts that you want to control via the PhoneDialog.
Before making an outgoing call, you can select which account is to be used at the top left of the PhoneDialog display.
The respective account is also displayed in the same place for an incoming call.
Under [Options > PhoneDialog (TAPI device) > Display as] you can assign short names to the SIP accounts.
Unfortunately, it is not intended to link the SIP accounts with several PhoneDialogs. Calls would then always be displayed in both PhoneDialogs.
The SIP SoftPhone should therefore only be assigned to one PhoneDialog.
Under [Options > PhoneDialog (TAPI device) > Driver configuration > Edit] you can select a conference mode:
Konferenzschaltung: Ab Version 3.1.025 unterstützt die SIP-SoftPhone-Komponente eine integrierte Konferenzschaltung.
- personal conference room: If you have a personal conference room in the telephone system, you can enter the conference room number here.
One conference room is required for each CTI Client user. The conference room must not require a PIN input. It would therefore be advisable to use a longer (secret) conference room number. These conference rooms should also not require name announcements.
- 3-Party-Conference via DTMF *3: Customers of "Deutsche Telefon Standard AG" can use this mode. The conference function is also to be activated by "Deutsche Telefon Standard AG".
Hint: Under the [SIP Service Provider > Settings], the DTMF mode should be set to "RFC 2833".
- A prerequisite for switching a three-way conference is that you have already set up a consultation call (a held call and an active call).
SIP Service Provider - Settings
A SIP service provider can be:
- A local PBX that supports SIP extensions.
- A cloud VoIP phone system.
Name: internal display name
IP-Address: DNS name or IP address of the VoIP telephone system
- fritz.box or 192.168.0.10
UDP / TCP: Type of SIP connection between CTI Client and telephone system.
- UDP: SIP control packets are transmitted/routed individually. (Standard)
- TCP: SIP control packets are transmitted over a previously established TCP connection.
Proxy: If your company network is secured via a proxy and the SIP control also has to run via this proxy, then the IP address of your company proxy must be entered here. May ask your network administrator for help.
NAT and Firewalls
These settings may be relevant if you are using a cloud phone provider and your local network is connected to the outside world via a NAT router (one external IP address).
If the connection to your cloud telephone provider does not work, you could test whether it works with the option [Evaluate parameter'"received"] or [Use STUN server].
Via a STUN server the CTI Client can determine your external router IP.
The telephone providers usually provide their own STUN server.
Register Interval: Determines how many seconds the SIP registration to the telephone system is renewed.
Keep-Alive interval: Determines how many seconds a small keep-alive packet is sent to the telephone system. The keep-alive packets are used to keep the UDP connection to the telephone system open in your router/firewall.
SIP session timer: Determines how many seconds active calls are confirmed with a RE-INVITE.
Use Connected UDP socket: If calls appear in the CTI Client that do not come from your VoIP provider/telephone system, this option could solve the problem with the fake calls.
All UDP packets that do not come from your VoIP provider/telephone system are ignored.
You can access this configuration dialog via:
[Options > PhoneDialog (TAPI Device) > Driver Configuration > Edit > SIP Service Provider > Settings > Tab "Features"].
Enter the pickup code of your telephone system here.
- For a pickup group, you may only need to enter the pickup code. Frequently *8 is used for pickup.
- For direct pickup, the extension number must also be dialed in (eg *8234). In this case, the hook is to be set at: "+ Phone number".
The "Direct Pickup" is done via the right-click menu of the speed dial buttons or via the pickup tool button from the PhoneDialog.
- In some telephone systems the pickup works without a pickup code. When the extension rings, pickup is done by simply calling the extension. In this case, uncheck "Pickup code". However, the tick next to "+phone number" must be checked.
- If the SIP-User-ID has to be dialed instead of the phone number for pickup, you can enter this separately, for example, in the speed dial buttons.
From version 3.1.008 two transfer modes are supported. The details of the A and C subscribers relate to the transfer of an incoming call.
- REFER C to A: The caller is connected to the consultation call.
- REFER A to C: The consultation call is connected to the caller.
If you have a call on hold and an active consultation call and then the transfer does not work, please set the alternative transfer mode here.
Call Forward / Do Not Disturb
Call forwarding and call protection can be switched via the icons at the bottom right of the orange PhoneDialog display.
Programming directly in the telephone system:
Vorteil: Rufumleitung/Anrufschutz sind auch aktiv, wenn der CTI Client geschlossen ist.
- Mode 1: about Control Codes
- The corresponding codes are to be configured on the tab "Functions".
- To set call forwarding, the CTI Client dials the corresponding code + phone number.
- The CTI Client notes whether call forwarding / do-not-disturb is set and shows it again after a restart.
- Mode 2: via CSTA/SUBSCRIBE (e.g. used by BroadSoft and Peoplefone)
- CSTA commands are sent to the PBX system using SIP-SUBSCRIBE.
- The PBX system informs the CTI Client via SIP NOTIFY that call forwarding/do-not-disturb has been set.
Otherwise, the CTI Client will handle call forwarding / call protection.
Advantage: The diverted/blocked calls are entered as missed in the journal.
Disadvantage: Call forwarding / call protection are only active when the CTI Client is running.
Active call forwarding / call protection applies to all registered SIP accounts. If several SIP accounts are active, but a call forwarding should only be set for one account, then you can assign a speed dial key with the corresponding programming code.
CLIR (phone number suppression)
If necessary, the transmission of your own telephone number for a call can be suppressed by means of a code prefix. Enter the relevant CLIR code in the "CLIR" field. Then the CLIR switch is available in the PhoneDialog.
You need the DTMF postdialing, for example, for voice announcements with menu selection.
"RFC 2833" (default): The tones are transmitted as digital commands in the audio channel.
Inband: The tones are generated directly by the CTI Client and transmitted in the audio channel.
SIP-Info: The sounds are generated by your telephone system.
Outside Line Access Code
If you have configured accounts from different SIP providers and only one SIP provider requires an outside line, you can optionally set an outside line code. This is usually a 0.
Amtsholung wird dann automatisch vorgewählt.
And the outside line prefix is removed from the phone numbers reported by the telephone provider.
Warning: In this case the outside line prefix must not be entered under [Options > PhoneDialog (TAPI Device) > Dial Settings > PBX System]!
The "Extended Settings" apply to all SIP accounts.
First port used locally
For the SIP control, the CTI Client uses the specified port. The default is port 5060. If the port is occupied, the next free port is used.
For audio then the next available ports are used.
- If you are using an external VoIP provider (cloud) and the internal parties (CTI Client to CTI Client) do not hear the each other:
- Configure a different local port for each CTI Client. Each CTI Client should have its own port area (10 ports).
Example: CTI Client 1: local port = 6000; CTI Client 2: 6010; CTI Client 3: 6020
Preferred audio codec
For now, only the A-Law and U-Law codecs are supported.
- If your audio connection is very noisy (this can occur during a consultation call, for example), then please set to "A-Law" and tick "Only support this codec".
Reject second call
If you are on a call, you can force other callers to receive a "busy" signal. The calls are still briefly displayed and entered in the journal.
- However, if the call is ringing on a desk phone at the same time, the caller may not get a busy signal. Here the telephone system would have to be configured accordingly.
Allow auto-answer via INVITE header
This option can be used in connection with the "TAPI for Asterisk" driver, for example. If dialing is made via the TAPI driver, the CTI Client SoftPhone is called first, which then automatically accepts the help call. The Asterisk telephone system then calls the destination number.
Scan specially formatted caller names
Example: If the telephone system transmits the called number (CalledID) in the caller name for an incoming call, this can be assigned to the TAPI parameter "CalledID". Thus, the CalledID can be displayed and logged separately.
When answering the call, turn the direction by RE-INVITE
For example, if you use a mobile hotspot, audio data may not be transmitted after accepting an incoming call. If you accept the call within 3 seconds, the audio works, if you accept the call later, it doesn't work.
Enabling this option can solve the problem.
Block Terminal- and RDP-Sessions
Example: You are using the CTI Client on your office PC as a SoftPhone. If you sometimes connect to your office PC from your home office via RDP, the remote audio quality may be too bad for making phone calls. When starting the CTI Client, a "TAPI device could not be opened" should appear.
Write log file (sip.log)
The "sip.log" log file can be activated for analysis purposes in the event of problems.