updateSipPhone.Rd
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 update information of a specific SIP Phone on a Zoom account.**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
updateSipPhone( phoneId, domain, register_server, transport_protocol = NULL, proxy_server, register_server2, transport_protocol2 = NULL, proxy_server2, register_server3, transport_protocol3 = NULL, proxy_server3, registration_expire_time = NULL, user_name, password, authorization_name, voice_mail, return_response = F )
phoneId | Unique Identifier of the SIP Phone. This can be retrieved from the List SIP Phones API. |
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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. |
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. |
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. |
voice_mail | The number to dial for checking voicemail. |
return_response | Whether to return the response instead of the response content. Defaults to FALSE. |