HELP CENTER

Provisioning


Login Fields

This field determines the set of options shown to the user when first setting up the SIP account. There are four possible options and the resulting screens that will be shown to the user for each option are shown below.

The most basic option is just to show the username and password which will be enough in many cases. The second option is for those VoIP providers or PBX SIP servers where an additional Authorization username is required.

If multiple subdomains need to be supported there is also the possiblilty to allow the user to enter the domain given to them by their provider. Again there is the option of entering the Authorization username. The screens shown to the user in each case are shown below.

Operator Code

For the Generic and Brands products an Operator Code is required. This is what the user will enter on first time setup to provision the softphones.

Provider Name

This is used on the Login screen and Dialer screen.

Domain

Server IP address or domain of your server. If port is other than 5060 specify the port as follows. provider.com:5065

Transport

Choice of UDP, TCP or TLS. TLS is only available in the premium package which has the encryption option.

DTMF Mode

Choice of Inband, RFC2833 and SIP INFO.

Wifi Codecs

Use the drop down select menu to select the codecs you wish to have available in your custom softphone. The codec order can also be set here with preferred codecs being left most in the horizontal list

3G Codecs

Use the drop down select menu to select the codecs you wish to have available in your custom softphone. The codec order can also be set here with preferred codecs being left most in the horizontal list

Global IP WiFi & Global IP 3G

If enabled the public IP address is used by the SIP client. There are separate setting for Wifi and cellular networks

STUN Enabled WiFi & STUN Enabled 3G

If enabled the public IP address is used by the SIP client in the SDP. There are separate setting for Wifi and cellular networks

STUN Server

Domain or IP address of STUN server in format stun.example.com:3478

Voicemail Number

The value entered here will be used when the 'Dial Voicemail' button is tapped in the mobile SIP client

Proxy

Enter Proxy in format proxy.com:5065

Register Expiry

This is the interval in seconds that the application will attempt to REGISTER with your provider.

UDP Keepalives

Enables sending of UDP keepalive packets. These are used to keep the pinhole open on your NAT/Firewall router so you receive incoming calls reliably.

SRTP

Enable Audio Encryption by default

SMS

If 'Enabled' show SMS Messaging in Tab Bar menu else it is hidden and unavailable for use.

Show Encryption Options Menu

The Audio Encryption menu and Transport options will visible to users. This will allow encryption to be enabled and disabled by the user

Show Log Menu

Logging option will be available to user. The log can then be enabled to your Support team to assist debugging.

Enable G729

The G729 codec will be available in WiFi & 3G Codecs menu. Please note there is a one time charge of $5 USD per user

Balance URL

To enable balance display enter the URL here. An example URL and response expected by the SIP client is shown below. The error tag is optional. The xml balance tag name is configurable. The outer response tag is mandatory. Also please note you must provide an https (SSL) url.

https://example.com/xml2.php?username=%user%&password=%pwd%

<response>
     <error>0</error>
     <balance>$10.00</balance>
</response>

Balance Tag

Used to specify name of xml tag which displays account balance.