Update SIP phone

PATCH

/sip_phones/{phoneId}

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 team.
Scope: sip_phone:write:admin

Rate Limit Label: Light

请求示例

Shell

JavaScript

Java

Swift

curl --location --request PATCH 'http://api.zoom.us/v2/sip_phones/' \
--form 'authorization_name=""' \
--form 'domain=""' \
--form 'password=""' \
--form 'proxy_server=""' \
--form 'proxy_server2=""' \
--form 'proxy_server3=""' \
--form 'register_server=""' \
--form 'register_server2=""' \
--form 'register_server3=""' \
--form 'user_name=""' \
--form 'voice_mail=""'

响应示例

200 - 示例 1

{}

请求参数

Path 参数

phoneId

string

必需

Unique Identifier of the SIP Phone. This can be retrieved from the List SIP Phones API.

Body 参数multipart/form-data

authorization_name

string

必需

Authorization name of the user registered for SIP Phone.

<= 64 字符

domain

string

必需

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

<= 64 字符

password

string

必需

The password generated for the user in the SIP account.

proxy_server

string

必需

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.

proxy_server2

string

必需

proxy_server3

string

必需

register_server

string

必需

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.

register_server2

string

必需

register_server3

string

必需

registration_expire_time

integer

可选

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.

>= 1<= 127

默认值:

示例值:

transport_protocol

enum<string>

可选

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

枚举值:

UDPTCPTLSAUTO

transport_protocol2

enum<string>

可选

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

枚举值:

UDPTCPTLSAUTO

transport_protocol3

enum<string>

可选

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

枚举值:

UDPTCPTLSAUTO

user_name

string

必需

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

<= 64 字符

voice_mail

string

必需

The number to dial for checking voicemail.

<= 255 字符

返回响应

🟢200**Error Code:** `200` Permission missing: Enable SIP Phone Integration by contacting a Zoom Admin first.

application/json

Body

object {0}

🟢204**HTTP Status Code:** `204` SIP Phone information updated successfully.

🔵300**Error Code:** `300` SIP phone not found: phoneId. The SIP phone with this phoneId was not found on this account.

🟠400**HTTP Status Code:** `400` Bad request.

修改于 2024-04-24 09:06:48

Delete SIP phone

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