03.5.3. Handset Fields

Handset Fields

Extension – This is the extension number you wish to assign to the new handset. The Extension number must be unique. Noojee Provision will automatically assign a free handset.

Username – The Username is used by the handset to identify itself to the system. This normally should be a numerical entry(depending on the type of handset). We recommend using the extension number as the Username. The user does not normally need access to the username.

If Asterisk is acting as a SIP client to a remote SIP server that requires SIP INVITE authentication, then this field is used to authenticate SIP INVITEs that Asterisk sends to the remote SIP server. Also, for peers that register with Asterisk, this username is used in INVITEs until we have a registration.

Secret (password) – This is nearly always a numerical password, depending on the type of handset used. The Secret is used in conjunction with the Username to authenticate the handset to the Asterisk PBX. The user does not normally need access to the Secret.

If Asterisk is acting as a SIP client to a remote SIP server that requires SIP INVITE authentication, then this field is used to authenticate SIP INVITEs that Asterisk sends to the remote SIP server.

Context – The context controls which dial plan context is used when the handset attempts to make an outbound call. This should normally be set to 'default' (without the quotes).

Mailbox – Set the Mailbox to link the extension to a specific mailbox to enable the handset's message waiting indicator (when supported by the handset). The mailbox entry will usually be the same as the extension number.

Account Code – Users may be associated with a particular account code. Account codes can be used to group a set of extensions for such purposes as departmental level cost reporting. A number of the supplied reports use the Account Code to group call detail records. Once an account code is assigned all new Call Detail Records (CDR) will have the account code applied. Old records will not be affected.

Caller ID – If set the Caller ID is displayed when calling another extension within the organization. The Caller ID is usually set to the a persons name.

Pickup Group – Defines the set of 'Call Groups' this extension can pick up calls from. If an extension is ringing which belongs to a 'Call Group' and this handset has listed in its Pickup Group then this extension can can pickup the call by dialing *8.

A Call Group is a integer value in the range 0-63. Multiple Call Groups defined in a comma separated list. A range of call groups can be defined using an lower and upper bound separated by a hyphen, e.g. 1,9-11

Allows the handset to pick up calls from handsets in the call groups 1, 9, 10 and 11.

Call Group – A call group is used to place a handset into a call group (don't confuse this with a ring group). Once in a call group a ringing handset can be answered from any extension (by dialing *8) which is permitted to to perform a call pickup for that group (see Call Pickup Group).

A call group is an integer value in the range 0-63.

The remaining fields are for advanced users only and should not normally need to be adjusted.

Host Type – The handset is able to register its own IP Address if you enter the Host Type as “dynamic”.

If you wish to manually enter the handset's hostname enter the Host Type as “hostname”. If you intend to create an alternate IP Address for advanced security on a network that exceeds the normal limits of an internal network, you may wish to hardcode your own designated IP Address. To do this you will need to enter the Host Type as “IPAddr”. If you choose IPAddr ensure you create the IP Address in its field below.

Type – A type of registration with the server. The three domain options are as follows. “user” is for handset to register with a server, “peer” used for a server to register with another server, and “friend” is for calls registering both directions between multiple servers.

IP Address – This is the IP of the phone. Finding a phones IP varies depending on the type of phone. Use the phone's user manual to locate its IP address.

AMA Flags – An advanced function used for billing purposes by some billing systems. Leave the field blank or enter 'Default'.

• default: Sets the system default.

• omit: Do not record calls.

• billing: Mark the entry for billing

• documentation: Mark the entry for documentation.

 

Can Re-Invite – update|yes|no : If the handset is able to support SIP re-invites. Default yes.

Default IP – Default IP address of client host= is specified as DYNAMIC. Used if client has not been registered at any other IP address. Valid only for type=peer.

DTMF Mode – inband|info|rfc2833 : How the client handles DTMF signaling.
Default rfc2833.

From User – Specify user to display in the Caller ID when placing the call through an external SIP server. Valid only for type=peer.

From Domain – Sets default From: domain in SIP messages when placing calls _to_ peer. Valid only for type=peer.

Full Contact – SIP URI contact for realtime peer. Valid only for realtime peers.

Insecure – very|yes|no|invite|port : Specifies how to handle connections with peers. Default no (authenticate all connections).

Language – A language code defined in indications.conf - defines language for prompts.

MD5 Secret – Not currently supported.

NAT – involves re-writing the source and/or destination IP Addresses. yes|no : This variable changes the behavior of Asterisk for clients behind a firewall. This does not solve the problem if Asterisk is behind the firewall and the client on the outside. Values: yes|no|never|route. Default no which really means to use rfc3581 techniques.

Deny – IP address and network restriction.

Permit – IP address and network restriction.

Mask – IP address and network restriction.

Port – SIP port of the client.

Qualify – When the handset is turned on, the system then checks if it is still on at a designated interval. The default interval is 2,000 milliseconds.

RTP timeout – Terminate call if x seconds of no RTP activity when we're not on hold.

RTP Hold Timeout – Terminate call if x seconds of no RTP activity when we're on hold (must be larger than rtptimeout).

Disallow – Used in conjuction with 'Allow' to control what codecs are enabled. The Disallow field is applied first and then any 'Allow'ed codecs are added. The Default setting is 'all'.

Music on Hold – Allows control over which music on hold settings are used. The list of settings is held in musiconhold.conf.

Registration Interval – The time, in seconds, between each re-registration.

Registered Extension – Advanced feature for use by Asterisk engineers only.

Can Call Forward – Default yes. Domain yes|no. This is a server side setting. Most phones handle call forwarding themselves and as such this setting has no affect in these cases.

Set Variable – value : Channel variable to be set for all calls from this peer/user.