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Interconnection Solution Between Huawei U1900 and Polycom SoundStation IP Device

Publication Date:  2019-04-10 Views:  19 Downloads:  0
Issue Description

Huawei PBX: U1911 V200R003C30SPC500

Polycom SoundStation: IP device

Solution

Configuration Process

1. Configure SIP user numbers on the U1900 to allocate numbers.

a. Choose user management > SIP user.

On the SIP User configuration page, click Setup. Set device type to Common terminal and device identification to the SIP registration account or user ID required by Polycom devices.

Note:


From V200R003C30SPC300 of the unified gateway, two additional parameters, namely Service password and Self-service login password, are provided for a user being created. You can configure Service password and Self-service login password on the web management system, or run the config system subscriber defaultpwd password password webpassword webpassword command in global config mode to configure the default service password and self-service login password for the user. If the passwords are not configured either using the command or on the web management system, the user fails to be created. If the passwords are configured both using the command and on the web management system, the passwords configured on the web management system prevail.


Click confirm, and click save data in the upper right corner of the web management system.

 

2. Configure interconnection and user data on the Polycom device.

The U1911 is configured with only account information for number allocation. The IP address of the Polycom device is unknown and needs to be configured on its web management system.

(1) Select the Simple Setup wizard. By default, the network configuration such as IP address and gateway settings has been completed.

Configure the SIP server (U1911 information), SIP external proxy (external communication), and account setting (number allocation configuration).


SIP server address

IP address of the SIP server, that is, the IP address of the unified gateway.

SIP server port

Port number of the SIP server.

Note:

 If the transport mode is set to TLS, the default port number is 5061.

 If the transport mode is set to UDP or TCP, the default port number is 5060.

Line Identification Address

SIP user ID used for authentication. Set this parameter to the SIP device ID configured on the unified gateway.

Authentication user ID

Account name provided by the VoIP service provider, which is similar to or is a phone number. Set this parameter to the corresponding user number configured on the unified gateway.

Authentication password

Password used for the Polycom device to register with the SIP server. Set this parameter to the authentication password for the SIP user configured on the unified gateway.


(2) Confirm in Settings that the SIP settings and the line account information have taken effect. If the settings do not take effect, modify the settings as follows:




Parameter Name

Description

Account Registraion

SIP user ID used for authentication. Set this parameter to the SIP device ID configured on the unified gateway.

Account name

Account name provided by the VoIP service provider, which is similar to or is a phone number. Set this parameter to the corresponding user number configured on the unified gateway.

Password

Password used for the Polycom device to register with the SIP server.

Address

IP address of the SIP server, that is, the IP address of the unified gateway.

Port

Port number of the SIP server.

Note:

 If the transport mode is set to TLS, the default port number is 5061.

 If the transport mode is set to UDP or TCP, the default port number is 5060.

 If the transport mode is set to DNS-SRV, the default port number is 0.

Transport

There are four transport modes including UDP, TCP, TLS, and DNS-SRV. The first three are SIP transport modes, with TLS being encrypted transport.

Note:

The value of this parameter must be the same as the transport type configured for the SIP trunk. If the SIP trunk is not used to route outgoing calls, this parameter is optional.

DTMF Type

There are four types including INBAND, by RFC2833, SIP INFO, and AUTO+SIP INFO.

If non-G.711 codec is used, set this parameter to by RFC2833.


3. Click submit. The system will restart for the modification to take effect. If the system is not restarted, manually restart it.

END