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Manager: Configuration Mode Field Descriptions > Line > SIP Line

Line | SIP Line

Usability
• These settings are not mergeable. Changes to these settings will require a reboot of the system.

Configuration Settings
• Line Number: Default = Automatically assigned.
By default a value is assigned by the system. This value can be changed but it must be unique.

• ITSP Domain Name: Default = Blank.

This field is used to enter the domain part of the SIP URI provided by the ITSP. For example, in the SIP URI
[email protected], the domain part of the URI is example.com. For outgoing calls the user part of the SIP URI is
determined in a number of ways:

• 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. 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 (User | SIP | SIP Name).

• For the destination of the call, the user part of the TO SIP URI is determined by the dialing short codes of the form
9N/N"@example.com" where N is the user part of the SIP URI.

• ITSP IP Address: Default = 0.0.0.0. Release 4.0 to 6.0.

This value is provided by the SIP ITSP. This address must not be shared by any other IP, IP DECT or SIP line in the system
configuration.

• Release 6.0+: Domain name resolution using the ITSP Domain Name value can be used instead. This requires a DNS
server and the system to be acting as a DHCP client. If that is the case, leaving the ITSP IP Address as 0.0.0.0 will cause the
system to attempt to resolve the domain name using DNS.

• Release 6.1+: This setting has replaced by ITSP Proxy Address on the Transport tab.

• Prefix: Default = Blank.

This prefix is removed from the called number on outgoing calls if present. See SIP Prefix Operation below.

• National Prefix: Default = 0.

This prefix is added to calls identified as not being international.

• Country Code: Default = Blank.

Set to match the local country code of the system location.

• International Prefix: Default = 00.

This prefix is added to calls identified as not being national.

• Send Caller ID: Default = None.

Select which value the SIP line should use for the original calling party ID when routing twinned calls.

• This setting is also used for forwarded calls. Note that the values on the System | Twinning tab override this if set. For
incoming calls to a hunt group, the hunt group details will be provided and not the details of the answering agent. This
setting is mergeable.

• The SIP line Send Caller ID setting takes priority.

• The values on the System | Twinning tab override the SIP lines Send Caller ID setting.

• Diversion Header
Use the information from the Diversion Header.

• Remote Party ID
Use the Remote Part ID.

• P Asserted ID
Use the contact information from the P Asserted ID.

• None
This option corresponds to the ISDN withheld setting.

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• Network Type: Default = Public.

This option is available if Restrict Network Interconnect (System | Telephony | Telephony) is enabled. It allows the trunk to
be set as either Public or Private. The system will return number unobtainable indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers, forwarding and conference calls.

• Due to the nature of this feature, its use is not recommended on systems also using any of the following other system
features: multi-site networks, VPNremote, application telecommuter mode.

• Association Method: Default = By Source IP Address.

This setting sets the method by which a SIP line is associated with an incoming SIP request.

• The match criteria used for each line can be varied. The search for a line match for an incoming request is done against
each line in turn using each lines Association Method. The order of line matching uses the configured Line Number
settings until a match occurs. If no match occurs the request is ignored. This method allows multiple SIP lines with the
same address settings. This may be necessary for scenarios where it may be required to support multiple SIP lines to the
same ITSP. For example when the same ITSP supports different call plans on separate lines or where all outgoing SIP
lines are routed from the system via an additional on-site system.

• By Source IP Address
This option uses the source IP address and port of the incoming request for association. The match is against the configured
remote end of the SIP line, using either an IP address/port or the resolution of a fully qualified domain name.

• "From" header hostpart against ITSP domain

This option uses the host part of the From header in the incoming SIP request for association. The match is against the ITSP
Domain Name above.

• R-URI hostpart against ITSP domain

This option uses the host part of the Request-URI header in the incoming SIP request for association. The match is against
the ITSP Domain Name above.

• "To" header hostpart against ITSP domain

This option uses the host part of the To header in the incoming SIP request for association. The match is against the ITSP
Domain Name above.

• "From" header hostpart against DNS-resolved ITSP domain

This option uses the host part of the FROM header in the incoming SIP request for association. The match is found by
comparing the FROM header against a list of IP addresses resulting from resolution of the ITSP Domain Name above or, if
set, the ITSP Proxy Address on the Transport tab.

• "Via" header hostpart against DNS-resolved ITSP domain

This option uses the host part of the VIA header in the incoming SIP request for association. The match is found by
comparing the VIA header against a list of IP addresses resulting from resolution of the ITSP Domain Name above or, if set,
the ITSP Proxy Address on the Transport tab.

• "From" header hostpart against ITSP proxy

This option uses the host part of the “From” header in the incoming SIP request for association. The match is against the
ITSP Proxy Address on the Transport tab.

• "To" header hostpart against ITSP proxy

This option uses the host part of the From header in the incoming SIP request for association. The match is against the ITSP
Proxy Address on the Transport tab.

• R-URI hostpart against ITSP proxy

This option uses the host part of the Request-URI in the incoming SIP request for association. The match is against the ITSP
Proxy Address on the Transport tab.

• REFER Support: Default = On.

REFER is the method used by many SIP device, including SIP trunks, to transfer calls. These settings can be used to control
whether REFER is used as the method to transfer calls on this SIP trunk to another call on the same trunk. If supported, once the
transfer has been completed, the system is no longer involved in the call. If not supported, the transfer may still be completed
but the call will continue to be routed via the system.

• Incoming: Default = Auto

Select whether REFER can or should be used when an attempt to transfer an incoming call on the trunk results in an outgoing
call on another channel on the same trunk. The options are:

• Always
Always use REFER for call transfers that use this trunk for both legs of the transfer. If REFER is not supported, the call
transfer attempt is stopped.

• Auto
Request to use REFER if possible for call transfers that use this trunk for both legs of the transfer. If REFER is not
supported, transfer the call via the system as for the Never setting below.

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• Never
Do not use REFER for call transfers that use this trunk for both legs of the transfer. The transfer can be completed but
will use 2 channels on the trunk.

• Outgoing: Default = Auto

Select whether REFER can or should be used when attempt to transfer an outgoing call on the trunk results in an incoming
call on another channel on the same trunk. This uses system resources and may incur costs for the duration of the
transferred call. The options available are the same as for the Incoming setting.

• Method for Session Refresh: Default = Auto.

The SIP UPDATE method (RFC 3311) allows a client to update parameters of a session (such as the set of media streams and
their codecs) but has no impact on the state of a dialog.

• RE-INVITE
Re-Invite messages are sent for session refresh.

• UPDATE
UPDATE messages are sent for session refresh if the other end indicates support for UPDATE in the allow header.

• Auto
UPDATE messages are sent for session refresh if the other end indicates support for UPDATE in the allow header. If UPDATE
is not supported, RE-INVITE messages are sent.

• Session Timer: 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.
Setting the field to On Demand disables the session timer.

• Media Connection Preservation: Default = Disabled.

When enabled, allows established calls to continue despite brief network failures. Call handling features are no longer available
when a call is in a preserved state. Preservation on public SIP trunks is not supported until tested with a specific service provider.

• In Service: Default = On.

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

• URI Type: Default = SIP. Release 6.2+

When SIP or SIPS is selected in the drop-down box, the SIP URI format is used (for example, [email protected]). When Tel is
selected in the drop-down box, the Tel URI format is used (for example, +1-425-555-4567). This affects the From field of
outgoing calls. The To field for outgoing calls will always use the format specified by the short codes used for outgoing call
routing. Recommendation: When SIP Secured URI is required, the URI Type should be set to SIPS. SIPS can be used only when
Layer 4 Protocol is set to TLS.

• 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 TCP trunks, if the TCP connection is disconnected the trunk will be 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.

• Call Routing Method: Default = Request URI.

This field allows selection of which incoming SIP information should be used for incoming number matching by the system's
incoming call routes. The options are to match either the Request URI or the To Header element provided with the incoming
call.

• Originator number for forwarded and twinning calls: Default = Blank.

This field can be used to set a originator number for forwarded and twinned calls when using any of the Send Caller ID options
above other than None. If exported or imported as part of a trunk template, this setting is not supported by IP Office Basic
Edition - Quick Mode mode systems.

• 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. Select one of the following options:

• System Default
Use the system's Default Name Priority setting (System | Telephony | Telephony).

• Favour 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 Favour Directory method.

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• Favour 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.

• Caller ID from From Header: Default = Off.

Incoming calls can include caller ID information in both the From field and in the PAI fields. When this option is selected, the
caller ID information in the From field is used rather than that in the PAI fields.

• Send From In Clear: Default = Off.

When selected, the user ID of the caller is included in the From field. This applies even if the caller has selected to be or is
configured to be anonymous, though their anonymous state is honored in other fields used to display the caller identity.

• User-Agent and Server Headers: Default = Blank (Use system type and software level).
The value set in this field is used as the User-Agent and Server value included in SIP request headers made by this line. If the
field is blank, the type of IP Office system and its software level used. Setting a unique value can be useful in call diagnostics
when the system has multiple SIP trunks.

SIP Prefix Operation

The prefix fields Prefix, National Prefix, Country Code and International Prefix are available with the SIP Line settings. These
fields are used in the following order:

1. If an incoming number (called or calling) starts with the + symbol, the + is replaced with the International Prefix.

2. If the Country Code has been set and an incoming number begins with that Country Code or with the International
Prefix and Country Code, they are replaced with the National Prefix.

3. If the Country Code has been set and the incoming number does not start with the National Prefix or International
Prefix, the International Prefix is added.

4. If the incoming number does not begin with either the National Prefix or International Prefix, then the Prefix is added.

For example, if the SIP Line is configured with prefixes as follows:

• Line Prefix: 9

• National Prefix: 90

• International Prefix: 900

• Country Code: 44

Number Received Processing Resulting Number

+441707362200 Following rule 1 above, the + is replace with the International Prefix 901707362200
(900), resulting in 900441707362200.

The number now matches the International Prefix (900) and Country
Code (44).Following rule 2 above they are replace with the National
Prefix (90).

00441707362200 Following rule 2 above the International Prefix (900) and the Country 90107362200
Code (44) are replaced with the National Prefix (90).

441707362200 Following rule 2 above, the Country Code (44) is replace with the 901707362200
National Prefix (90).

6494770557 Following rule 3 above the International Prefix (900) is added. 9006494770557

OPTIONS Operation
For Release 8.1 FP1, the sending of OPTIONS messages on a SIP line is controlled by a number of fields as follows:

• If either REFER Support or the UPDATE Supported values are set to Auto, then OPTIONS messages are sent.

• If neither REFER Support or the UPDATE Supported values are set to Auto, then:

• If the LAN interface's Binding Refresh Time is set, addition OPTIONS messages are not sent.

• If the LAN interface's Binding Refresh Time is 0, the sending of OPTIONS messages depends on the line's Check OOS

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setting. If Check OOS is enabled, OPTIONS are not sent if line registrations occur more frequently, otherwise OPTIONS
are sent.

© 2013 AVAYA
15-601011 Issue 9..01
2:09 PM, Monday, November 04, 2013
(html_sip_line.htm)

Performance figures, data and operation quoted in this document are typical and must be specifically confirmed in writing by Avaya
before they become applicable to any particular order or contract. The company reserves the right to make alterations or
amendments at its own discretion. The publication of information in this document does not imply freedom from patent or any
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Last Modified: 7/5/2013

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