Reference9:Phone/User/General

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Here, you enter the parameters for registering with the innovaphone PBX. These parameters are preset by the administrator of the telephone system. First, you must select the required protocol for registration. The options are H.323 or SIP. The respective parameters are automatically adjusted to the selected protocol.


Contents

H.323 protocol

Primary gatekeeper address

  • Here, you specify the IP address at which the first responsible gatekeeper can be reached.

Secondary gatekeeper address

  • There should be a further gatekeeper in the network in case the first gatekeeper is unavailable. You enter the IP address of this gatekeeper here.

NOTE: When the phone works with Gatekeeper Discovery (Gatekeeper identifier) as primary registration and a Secondary gatekeeper address is configured, this direct IP address will be used as backup. So when the Gatekeeper discovery fails the secondary gatekeeper address is used to register.


Local endpoint address

  • If several local IP addresses are available on the IP telephone, this setting enables you to specify the one to be used for communication with the gatekeeper.

Note: Since the local IP address was preset for the IP telephone using the IP routing table, the value is only necessary in exceptional cases.

Gatekeeper identifier

  • If several gatekeepers are to be active at one address, a particular gatekeeper is identified amongst them using the name entered here.

Number

  • Here, you specify the call number required for the registration. This number is only required if no name was specified.

Name

  • The name you enter here is only required for the registration if the number was not specified.

SIP protocol

Protocol
SIP/UDP
SIP signaling using UDP as transport protocol (RFC 3261).
SIP/TCP
SIP signaling using TCP as transport protocol (RFC 3261).
SIP/TLS
SIP signaling using TLS as transport protocol (RFC 3261).

Primary server address

  • IP address at which the responsible SIP server may be reached.

Secondary server address

  • There should be a further SIP server in the network in case the first one is unavailable.

Note: Since the local IP address was preset for the IP telephone using the IP routing table, the value is only necessary in exceptional cases.

Domain

  • Instead of the IP address, you can specify the domain of the provider, which is found after the @ of the URI.

CGPN

  • The calling party number to be used on outgoing calls.

User ID

  • The user ID, which corresponds to the part in front of the @ of the URI.

Note: The user ID can be numeric or alphanumeric.

STUN server

  • The IP address or domain name must be configured if the telephone uses a non-public IP address, but the server is accessible under a public IP address. The value is given by the SIP provider or administrator.

Protocol independent

The following parameters can be set protocol independently. These parameters are automatically negotiated during connection setup. A change is only necessary if the gatekeeper/server has special requirements regarding the connection protocol.

Username

  • In some SIP registrations, a separate user name is required for authorization. In all other cases, the field should be left empty.

Password

  • The registration requires a password, which can be agreed in this setting.

Retype

  • The confirmation prompt for the password.

Dial tones

  • In some countries, the dial tones are different. Here, you make the pre-settings for the specific country.

Enblock dialling timeout [s]

  • The IP telephone supports single digit dialling (overlapped sending), i.e. after a setup request the digits are sent as typed. After all digits of the number have been sent the connection is established automatically by the exchange. If the exchange does not support single digit dialling, you can switch to en-bloc dialling mode by entering a timeout value here. In en-bloc dialling mode the typed digits are gathered locally. With each digit the timeout is started. If there is no digit entered in the timout period a setup request containing the gathered digits is sent.

Note: Especially the IP150 phone does not allow entering the number without pick-up the headset. So it makes sense to use this timeout for enblock dialing at the IP150.


General Coder Preference

This configuration applies to outgoing calls.

Coder

  • The protocols for voice compression are listed in this parameter.

Framesize [ms]

  • The packet sizes are automatically negotiated with 60 ms. With this parameter, you can specify a different RTP packet size.

Silence compression

  • No outbound packets are transmitted from the IP telephone during silence (no conversation).

Exclusive

  • This parameter allows no negotiation, but accepts only the preset parameters.


Local Network Coder

This configuration applies to incoming calls and only if the remote RTP address is considered to be a local network address. See Reference:Configuration/IP/Settings for information on how to declare local network addresses.

Coder

  • The protocols for voice compression are listed in this parameter.

Framesize [ms]

  • The packet sizes are automatically negotiated with 60 ms. With this parameter, you can specify a different RTP packet size.

Silence compression

  • No outbound packets are transmitted from the IP telephone during silence (no conversation).

Enable Secure RTP

  • An SRTP encryption key is offered. If remote party also provides an SRTP encryption key, media streaming will be locked against wiretapping. (since V7)

No DTMF Detection

  • DTMF tones are sent in-band through the media channel but not as separate signalling messages (that is, they are not signalled at all, neither in-band nor out-band but sent as plain voice).
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