How to make a call via SIP/H.323 and RTSP

Making a Call

To call external devices or a server, enter the in the format which corresponds to the type of recipient. The UI will look similar to the cases when you enter the name of a regular user. Right above the filtered , you will see a new slightly darkened row. You can interact with it, just like with any other.

Supported devices and call string formats:

• (including those using DTMF dialing);
• ;
• .

Since is a hardware SIP terminal, use the for calls to it.

Automatic processing of URLs for meetings created on third-party services like Zoom, Lifesize Cloud, Cisco Webex, or GoToMeeting, for example:

• https://zoom.us/j/842858705

• https://call.lifesizecloud.com/10132060

• #sip:pr1630419186@meetingsemea590.webex.com

Using DTMF Tones

You can send DTMF commands to DTMF-compatible devices in RTP EVENT and SIP INFO modes. To learn more about DTMF commands, please read the documentation provided by the manufacturer for your device.

The following symbols can be used to add pauses directly to the call string:

• , — short pause (a few seconds)

• ; — long pause (waiting for a dial tone from the caller).

For example, if you want to call a SIP server with IP 192.168.1.99 from the client application to a conference protected by PIN 123456, you can avoid manual PIN entry by using a URI with a preset:

#sip:@192.168.1.99;123456

To call 13478783263 with extension 222, you can use the following call line:

#tel:13478783263,222