Call Forward / Do Not DisturbĬall forwarding and call protection can be switched via the icons at the bottom right of the orange PhoneDialog display. 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. REFER A to C: The consultation call is connected to the caller.REFER C to A: The caller is connected to the consultation call.The details of the A and C subscribers relate to the transfer of an incoming call. The "Direct Pickup" is done via the right-click menu of the speed dial buttons or via the pickup tool button from the PhoneDialog.įrom version 3.1.008 two transfer modes are supported. In this case, the hook is to be set at: "+ Phone number". For direct pickup, the extension number must also be dialed in (eg *8234).For a pickup group, you may only need to enter the pickup code.PickupĮnter the pickup code of your telephone system here. You can access this configuration dialog via: 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.Īll UDP packets that do not come from your VoIP provider/telephone system are ignored. SIP session timer: Determines how many seconds active calls are confirmed with a RE-INVITE. The keep-alive packets are used to keep the UDP connection to the telephone system open in your router/firewall. Keep-Alive interval: Determines how many seconds a small keep-alive packet is sent to the telephone system. Register Interval: Determines how many seconds the SIP registration to the telephone system is renewed. The telephone providers usually provide their own STUN server. Via a STUN server the CTI Client can determine your external router IP. If the connection to your cloud telephone provider does not work, you could test whether it works with the option or. 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). May ask your network administrator for help. 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. TCP: SIP control packets are transmitted over a previously established TCP connection.UDP: SIP control packets are transmitted/routed individually.UDP / TCP: Type of SIP connection between CTI Client and telephone system. IP-Address: DNS name or IP address of the VoIP telephone system ![]() A local PBX that supports SIP extensions.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). ![]() Hint: Under the, the DTMF mode should be set to "RFC 2833". The conference function is also to be activated by "Deutsche Telefon Standard AG".
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