Jitsi call sip control how to#

If you are using a headset, wiping your cheek, touching the microphone or moving the microphone will all be heard _loudly_ by everyone on the call.

Be prepared to adjust these levels at the start of the call. You can control the level of speaker output and also the level of your microphone input.

Know where your sound settings are for your operating system.

Also jingling keys or tapping fingers create sounds your microphone will pick up.

If you use your laptop microphone, please turn your cellphone/telephone to vibrate/silence and move it away from your laptop.

Be sure to test your microphone configuration before joining the call, sound drivers/cards and microphones need to work well together for us to hear you. Headsets and microphones will be closer to the source of the generated sound and will transmit a richer and more pleasant audio signal.

Use a headset or microphone which you have tested and which you are sure works.

Background noise will be picked up by your microphone and everyone on the call will hear it. There are some things you can do to increase the audio quality of the experience for yourself and all participants in the call. The quality of the audio experience directly effects the ability of other participants to understand you clearly. This type of account will let you make SIP calls from your computer without registering to any SIP server. Once you have Jitsi running, you must first create an account. The following clients have been tested with the system:

* - Play a prompt that describes the menu (then press a digit as described in the prompt)Īny SIP soft or hard phone should work fine.

This can be easily changed as needed.Īlternatively, if you are calling in via SIP, you can call a bridge directly: In-call menuīy default, conference users have the following menu available to them during a conference call: We have currently reserved 6000 through 7999 as conference bridges. Once you call into the system, it will ask you for a conference number. The following methods are available for calling the conferencing system:

Jitsi call sip control how to#

This page documents how to access the system.

The OpenStack Infrastructure team maintains a voice conferencing system based on Asterisk.