Zoom’s Phone System Integration PSI , also referred as SIP phones, enables an organization to leverage the Zoom client to complete a softphone registration to supported premise based PBX system. End users will have the ability to have softphone functionality within a single client while maintaining a comparable interface to Zoom Phone. Use this API to enable a user to use SIP phone.**Prerequisites**:* Currently only supported on Cisco and Avaya PBX systems. * The account owner or account admin must first enable SIP Phone Integration by contacting the Sales https://zoom.us/contactsales team. **Scope:** sip_phone:write:admin ** Rate Limit Label https://marketplace.zoom.us/docs/apireference/ratelimits#ratelimits :** Light

createSipPhone(
  domain,
  register_server,
  transport_protocol = NULL,
  proxy_server,
  register_server2 = NULL,
  transport_protocol2 = NULL,
  proxy_server2 = NULL,
  register_server3 = NULL,
  transport_protocol3 = NULL,
  proxy_server3 = NULL,
  registration_expire_time = NULL,
  user_name,
  password,
  authorization_name,
  user_email,
  voice_mail,
  return_response = F
)

Arguments

domain

The name or IP address of your provider’s SIP domain. example: CDC.WEB .

register_server

IP address of the server that accepts REGISTER requests. Note that if you are using the UDP transport protocol, the default port is 5060. If you are using UDP with a different port number, that port number must be included with the IP address.

transport_protocol

Protocols supported by the SIP provider. The value must be either UDP, TCP, TLS, AUTO.

proxy_server

IP address of the proxy server for SIP requests. Note that if you are using the UDP transport protocol, the default port is 5060. If you are using UDP with a different port number, that port number must be included with the IP address. If you are not using a proxy server, this value can be the same as the Register Server.

register_server2

IP address of the server that accepts REGISTER requests. Note that if you are using the UDP transport protocol, the default port is 5060. If you are using UDP with a different port number, that port number must be included with the IP address.

transport_protocol2

Protocols supported by the SIP provider. The value must be either UDP, TCP, TLS, AUTO.

proxy_server2

IP address of the proxy server for SIP requests. Note that if you are using the UDP transport protocol, the default port is 5060. If you are using UDP with a different port number, that port number must be included with the IP address. If you are not using a proxy server, this value can be the same as the Register Server, or empty.

register_server3

IP address of the server that accepts REGISTER requests. Note that if you are using the UDP transport protocol, the default port is 5060. If you are using UDP with a different port number, that port number must be included with the IP address.

transport_protocol3

Protocols supported by the SIP provider. The value must be either UDP, TCP, TLS, AUTO.

proxy_server3

IP address of the proxy server for SIP requests. Note that if you are using the UDP transport protocol, the default port is 5060. If you are using UDP with a different port number, that port number must be included with the IP address. If you are not using a proxy server, this value can be the same as the Register Server, or empty.

registration_expire_time

The number of minutes after which the SIP registration of the Zoom client user will expire, and the client will auto register to the SIP server.

user_name

The phone number associated with the user in the SIP account.

password

The password generated for the user in the SIP account.

authorization_name

Authorization name of the user registered for SIP Phone.

user_email

The email address of the user to associate with the SIP Phone. Can add .win, .mac, .android, .ipad, .iphone, .linux, .pc, .mobile, .pad at the end of the email ex. user@test.com.mac to add accounts for different platforms for the same user.

voice_mail

The number to dial for checking voicemail.

return_response

Whether to return the response instead of the response content. Defaults to FALSE.