SIP Line

Last Updated : Apr 21, 2016 |

Navigation: System Settings > Line > Add/Edit Trunk Line > SIP Line > SIP Line

Configuration Settings

These settings are mergeable with the exception of the Line Number setting. Changing the Line Number setting requires a “merge with service disruption”. When the configuration file is sent to the system, the SIP trunk is restarted and all calls on the line are dropped.

Offline editing is not required.

Field

Description

Line Number

Default = Auto-filled. Range = 1 to 249 (IP500 V2)/1 to 349 (Server Edition).

The line number must be unique for each line in the configuration. IP500 V2 systems reserved line numbers 1 to 16 for internal hardware.

ITSP Domain Name

Default = Blank.

This field is used to specify the default host part of the SIP URI in the From, To, and R-URI fields for outgoing calls. For example, in the SIP URI name@example.com, the host part of the URI is example.com. When empty, the default host is provided by the SIP Line > SIP Transport > ITSP Proxy Address field value. If multiple addresses are defined in the ITSP Proxy Address field, then this field must be defined.

For the user making the call, the user part of the From SIP URI is determined by the settings of the SIP URI channel record being used to route the call (see SIP Line > SIP URI > Local URI). This will use one of the following:

  • a specific name entered in Local URI field of the channel record.

  • or specify using the primary or secondary authentication name set for the line below.

  • or specify using the SIP Name set for the user making the call (Call Management > Users > Add/Edit Users > SIP > SIP Name).

For the destination of the call, the user part of the To and R-URI fields are determined by dial short codes of the form 9N/N"@example.com” where N is the user part of the SIP URI and "@example.com" is optional and can be used to override the host part of the To and R-URI.

Local Domain Name

Default = Blank.

An IP address or SIP domain name as required by the service provider. When configured, the Local Domain Name value is used in

  • the From and Contact headers

  • the PAI header, if Line > SIP Advanced is checked

  • the Diversion header

If both the  ITSP Domain Name and Local Domain Name are configured, Local Domain takes precedence.

Local Domain Name is not used in the Remote Party ID header.

URI Type

Default = SIP URI.

Set the format the IP Office uses for SIP URI entries in headers.

  • SIP URI - Use SIP URI format. For example, display <sip:content@hostname>

  • Tel - Use Tel URI format. For example, +1-425-555-4567. This affects the From field of outgoing calls. The To field for outgoing calls uses the format specified by the short codes used for outgoing call routing.

  • SIPS - Use SIPS format for all URIs. SIPS can be used only when Layer 4 Protocol is set to TLS.

Location

Default = Cloud.

You can set Location values for the IP Office system and for individual extensions and lines. Associating a line with a location:

  • Applies the location's call admission control (CAC) settings to the line

  • For SIP lines that support RFC4119/RFC5139, emergency calls using the line can include the location's address information.

Prefix

National Prefix

International Prefix

Country Code

The IP Office uses these values to adjust incoming numbers to match the format required for outgoing calls and used in system directory entries.

  1. If the number starts with a + symbol, the symbol is replaced with the International Prefix.

  2. If the Country Code has been set:
    1. If the number begins with the Country Code, or International Prefix plus Country Code, the IP Office replaces them with the National Prefix.

    2. If the number does not start with the National Prefix or International Prefix, the IP Office adds the International Prefix.

  3. If the incoming number does not begin with the National Prefix or International Prefix, the IP Office adds the Prefix.

Name Priority

Default = System Default.

For SIP trunks, the caller name displayed on an extension can either be that supplied by the trunk or one obtained by checking for a number match in the extension user's personal directory and the system directory. This setting determines which method is used by the line. The options are:

  • System Default: Use the system setting System | Telephony | Telephony | Default Name Priority.

  • Favor Trunk: Display the name provided by the trunk. For example, the trunk may be configured to provide the calling number or the name of the caller. The system should display the caller information as it is provided by the trunk. If the trunk does not provide a name, the system uses the Favor Directory method.

  • Favor Directory: Search for a number match in the extension user's personal directory and then in the system directory. The first match is used and overrides the name provided by the SIP line. If no match is found, the name provided by the line, if any, is used.

Description

Default = Blank. Maximum 31 characters.

You can use this field to enter a description for the configuration entry. The description is not used elsewhere.

Network Type

Default = Public.

This option is available when System | Telephony | Telephony | Restrict Network Interconnect  is enabled. It allows you to configure trunks as either Public or Private.

  • The IP Office will return number busy indication to any attempt to connect a call on a Private trunk to a Public trunk or the opposite.

  • The call restriction includes transfers, forwarding and conference calls.

  • Avaya does not recommended use of this feature on IP Office systems using any of the following features:  multi-site networks, VPNremote, application telecommuter mode.

In Service

Default = On.

When this field is not selected, the SIP trunk is unregistered and not available to incoming and outgoing calls.

Check OOS

Default = On.

If enabled, the system will regularly check if the trunk is in service using the methods listed below. Checking that SIP trunks are in service ensures that outgoing call routing is not delayed waiting for response on a SIP trunk that is not currently usable.

For UDP and TCP trunks, OPTIONS message are regularly sent. If no reply to an OPTIONS message is received the trunk is taken out of service.

For trunks using DNS, if the IP address is not resolved or the DNS resolution has expired, the trunk is taken out of service.

Session Timers

Field

Description

Refresh Method

Default = Auto.

The options are: Auto, Reinvite or Update.

When Auto is selected, if UPDATE is in the Allow: header from the far SIP endpoint, then it is used. Otherwise INVITE is used.

Timer (seconds)

Default = On Demand. Range = 90 to 64800

This field specifies the session expiry time.  At the half way point of the expiry time, a session refresh message is sent. When set to On Demand, IP Office will not send a session refresh message but will respond to them.

Redirect and Transfer

Redirection and blind transfer are configured separately. By default, they are disabled.

A supervised transfer occurs when a consultation call is made and the REFER contains a Replaces: header indicating the CallID of another call leg which the REFERing agent has already initiated with the REFER target.

Note:
  • Do not change these settings unless directed to by the SIP service provider.

Field

Description

Incoming Supervised REFER

Default = Auto.

Determines if IP Office will accept a REFER being sent by the far end. The options are:

  • Always: Always accepted.

  • Auto: If the far end does not advertise REFER support in the Allow: header of the OPTIONS responses, then IP Office will reject a REFER from that endpoint.

  • Never: Never accepted.

Outgoing Supervised REFER

Default = Auto.

 Determines if IP Office will attempt to use the REFER mechanism to transfer a party to a call leg which IP Office has already initiated so that it can include the CallID in a Replaces: header. The options are:

  • Always: Always use REFER.

  • Auto: Use the Allow: header of the OPTIONS response to determine if the endpoint supports REFER.

  • Never: Never use REFER.

Send 302 Moved Temporarily

Default = Off.

A SIP response code used for redirecting an unanswered incoming call. It is a response to the INVITE, and cannot be used after the 200 OK has been sent as a response to the INVITE.

Outgoing Blind REFER

Default = Off.

When enabled, a user, voicemail system or IVR can transfer a call by sending a REFER to an endpoint that has not set up a second call. In this case, there is no Replaces: header because there is no CallID to replace the current one. This directs the far end to perform the transfer by initiating the new call and release the current call with IP Office.