Who needs to read this?
You want to manually configure Lumicall to connect to Kamailio, OpenSIPs, reSIProcate/repro, FreeSWITCH, Asterisk or another SIP server.
How to access the account settings
On the registration screen, you can register with Lumicall, or you can click the link at the bottom of the form to setup manually.
Once the registration form is gone, you can run the main Lumicall program by clicking the Lumicall icon.
When you have the Lumicall dialer screen open, press the menu button
Press SIP Identities - you will see a list of any existing identities/accounts, or a blank screen if none exist.
Here, you have three choices:
- Press `menu' and `add' to add a new identity/account
- Click an existing identity to edit the settings
- Long-press an existing identity to delete it (you will be prompted to confirm)
We recommend that you use TLS (not UDP or TCP) for SIP. This is quite important, because:
- large SIP messages can be truncated if transmitted using UDP (this is a common problem when using ICE/TURN, because the SDP data is very long: Lumicall has a full ICE agent)
- SIP messages can be mangled by NAT routers if using UDP or TCP, giving unpredictable results. The NAT router is trying to help, but sometimes it messes up the packet.
- TLS eliminates all these risks and increases the probability of successful connection
We recommend that you use DNS SRV records for your SIP identity. If your SIP identity is firstname.lastname@example.org, you should create DNS SRV records:
- _sips._tcp.example.org pointing at your SIP server
- _stun._udp.example.org pointint at your STUN/TURN relay server
If you do this, you do not have to manually configure the register server name, outbound proxy name or STUN server name
- SIP address/URI: this will be used as the `From' header in SIP communication
- Profile enabled: remove the tick, and the profile will not be able to receive calls
- Security mode: you can choose ZRTP (recommended), SRTP (necessary for some networks) or none
- Gateway to PSTN: tick this box, and when you dial a number, your phone will offer you the choice to send the call using this SIP account
- Intl. dialing prefix: If you dial a number with a leading plus (e.g. +44 20 7000 8000), Lumicall will remove the plus and replace it with the prefix when using this account. For example, if you enter the value 00, the number will be dialed 00442070008000
- Authorization username and password: these will be used for SIP and for STUN/TURN authentication
- Registration: this must be ticked if you want to receive calls
- Registration/outbound proxy/STUN server names: only needs to be filled in if you are not using DNS SRV records
- Registration/outbound proxy/STUN server protocol/port values: only needs to be filled in if you are not using DNS SRV records. These values are ignored if DNS SRV records are used.
- Use outbound proxy: it is recommended that you tick this box, it will cause all calls to be routed via the proxy server for your own SIP domain. If the box is not ticked, then Lumicall will try to call directly to the callee's proxy server: this often doesn't work directly from a phone, because many proxy servers expect third-party calls to be authenticated with an X.509 certificate
- Use STUN/TURN protocol: this is highly recommended as it ensures the phone will be able to make connections through NAT routers.