Recently we have integrated our demo server with Asterisk and now it's possible to call to demo room from SIP phone installed on your mobile device.
We mostly deal with Linphone, however you can try any other SIP phone for this.
To try SIP integration with OpenMeetings you would need:
- Register on the server
- Install and configure SIP phone on your mobile
- If you are under firewall, make sure that outgoing SIP port (5060) and RTP ports (10000-20000) are opened
- Log into OpenMeetings in your browser and enter the "SIP Testing Room" (for this room SIP integration is turned on on the server)
- Call "40021" (this is a number of the SIP Testing Room) from your mobile device
That's all - now you can participate in OpenMeetings conference using your mobile!
You need to set the following SIP phone parameters on your mobile:
- Login: your OpenMeetings login on demo.dataved.ru
- Password: your OpenMeetings password
- Domain: demo.dataved.ru
- Realm: asterisk
- Trasnport: UDP
- "Enable video" switch should be turned off
- If you are unable to register in your Linphone, login into OpenMeetings, goto your personal settings and re-save your password. Then try to register via Linphone again.