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Advanced options: Interaction SIP stations or templates

The following tables describe the advanced options for Interaction SIP Station phones or templates. For information on how to access these settings, see Configure managed IP phones or templates.

Note: These options are used only for diagnostic and troubleshooting purposes. Do not change them without clear direction from a PureConnect Customer Care representative.

Provisioning

Option	Description	Default
Provisioning Method	This setting supports several methods for determining the provisioning server to which the IP phone should connect. The options are DHCP Options, Disabled, Static URL. Depending on which method is selected, the DHCP Option or the Static URL may be required below.	DHCP Options
DHCP Option	This option specifies the Dynamic Host Configuration Protocol (DHCP) option number that identifies the vendor and functionality of a DHCP client.	160
Static URL	Specify the CIC server by entering the host name or IP address. All IP addresses in managed phones configuration are pass-through strings to support IPv6. Note: If Provisioning Method is set to Disabled, the Interaction SIP Station does not attempt to connect with the provisioning server. Therefore configuration of firmware is not requested, so that configuration information requires updates through the web interface.	blank or <empty>

Syslog Tracing

The following are all of the Syslog tracing options for Interaction SIP stations. The available options depend on the model.

Option	Description	Default
Syslog Enabled	This setting enables or disables diagnostic logs for Interaction SIP Stations in the form of Syslog messages. Setting this to Yes requires a Syslog Server to be specified. Use this only at the direction of an authorized PureConnect Customer Care engineer.	No
Syslog Server	This setting specifies the IP address of a third party Syslog server used to capture diagnostic logs and error messages generated by the Interaction SIP Station. A valid Syslog server IP address is required if Syslog Enabled is set to Yes. All IP addresses in managed phones configuration are pass-through strings to support IPv6 in a future release.	The default IP address is 0.0.0.0, which is invalid until set to a valid address.
Server Port	This setting defines the UDP port of the Syslog Server. The valid range is from 0 to 65,535.	514
Voip Application	This setting specifies the level at which Syslog messages are generated related to VoIP. Select from the following list: None Emergency Error Warning Notice Info Deb	None
Control Center	This setting specifies the level at which Syslog messages are generated related to Networking	None
LCD Display	This setting specifies the level at which Syslog messages are generated related to LCD Display and other keypresses.	None
Web	This setting specifies the level at which Syslog messages are generated related to phone web server.	None
Watchdog	This setting specifies the level at which Syslog messages are generated related to the watchdog process, which keeps other processes running.	None
802.1x	This setting specifies the level at which Syslog messages are generated related to the security protocol.	None
Kernel	This setting specifies the level at which Syslog messages are generated related to the kernel process of the phone.	None
DSP	This setting specifies the level at which Syslog messages are generated related to the voice engine of the phone.	None

Gain

The microphone settings adjust the volume for the benefit of the caller and the speaker settings adjust the volume for the benefit of the agent. We recommend you talk to authorized PureConnect Customer Care engineers if you have problems with audio volume.

Digital settings adjust volume to the line while the analog settings adjust volume to headset. Select the intended value from the drop-down list.

Option	Description	Default
Headset Digital Microphone Gain (default +0 dB)	This setting controls the gain of the RTP audio stream sent from the Interaction SIP Station device to the line.	+0 dB
Headset Digital Speaker Gain (default +0 dB)	This setting controls the gain of the RTP audio stream from the line through the Interaction SIP Station device and to the headset.	+0 dB
Headset Analog Microphone Gain (default +39 dB)	This setting controls the audio signal gain from the analog headset microphone into the Interaction SIP Station device.	+39 dB
Headset Analog Speaker Gain (default -12 dB)	This setting controls the audio signal gain from the Interaction SIP Station device to the analog headset.	-12 dB
Headset Analog Sidetone Gain (default - 12 dB)	This setting controls the sidetone audio signal gain from the Interaction SIP Station device to the analog headset. The sidetone controls the agents ability to hear their own voice in the headset as they speak.	-12 dB

VLAN

Option

Description

Default

VLAN Discovery Mode

This setting specifies the method that this Interaction SIP Station uses to find its Virtual Local Area Network (VLAN). This setting controls the use of VLAN tagging (802.1Q) for audio streams over a VLAN:

Automatic (CDP+LLDP) – This (default) option uses CDP (Cisco Discovery Protocol) together with LLDP (Link Layer Discovery Protocol) to automatically obtain the VLAN ID assignment. This option allows for discovery using whichever mode is active, and to use VLANs on different switches without changing configuration of the phone.
Automatic (CDP) - This option uses CDP (Cisco Discovery Protocol) to automatically obtain the VLAN ID assignment. Use this option to strictly limit VLAN discovery to Cisco switches.
Automatic (LLDP) - This option uses LLDP ( Link Layer Discovery Protocol) to automatically obtain the VLAN ID assignment. Use this option to strictly limit VLAN discovery to non-Cisco switches.
Disabled – This option disables the use of VLANs which stops VLAN tagging for this Interaction SIP Station. If you do not have the standard Voice/Data VLAN set up in your network, using this option skips the VLAN discovery mode which causes the IP phone to start-up faster on a reboot.
Manual – This option requires you to specify a VLAN ID in the Manual VLAN ID field. This option allows the IP phone to generate traffic on the specified VLAN without having either CDP or LLDP protocols enabled on the network.

Manual VLAN ID

If the VLAN Discovery Mode field is set to "Manual", you can specify the VLAN ID for this Interaction SIP Station in this field by entering the value it will use. Valid values are 1 through 4094.

Note: Important information regarding Manual VLAN ID
If you configure an Interaction SIP Station (ISS) to a Manual VLAN ID, the phone will not listen to packets that come into its network port unless the appropriate 802.1q tagging is set in the header. Some network switches have been observed to strip away this header before sending the packets to the ISS in the field. If this occurs, and Manual VLAN ID is set on the device, the device will no longer hear any traffic and cannot be accessed. If this occurs, there are only two ways to recover the device:

1.The switch must be changed to not strip the 802.1q header. The simplest way to do this is to configure the switch port as a trunk port. The method for performing this is dependent on your type of switch.

2.The device must be reset to factory defaults to reset/clear the Manual VLAN ID value. This however is only possible if the phone is already running on the 1.2.2 version of firmware.

WARNING
Because of this danger of causing the device to be inaccessible, systems running on firmware versions prior to 1.2.2 will not honor the Manual VLAN ID value until you set a safety server parameter. The server parameter is "Provision ISS Manual VLAN ID Enabled" and needs to be set to "Yes" to enable it. Make sure you test one single phone to ensure your switch is not stripping the 802.1q VLAN tagging before you set any more phones!

If you choose to set the VLAN ID manually on an ISS and they are on pre-1.2.2 firmware, you will have to set your network switch ports to trunk ports in the case that they are stripping the 802.1q headers. If the ISS is on post 1.2.2 firmware, you will have to either configure the network switch ports as trunk ports, or perform the factory reset sequence on the device that is added in the 1.2.2 firmware. Be sure to check the support site for special notices.

LAN

Option	Description	Default
IP Assignment Mode	This setting specifies the method that this Interaction SIP Station uses to obtain an IP address: Automatic (DHCP) – (Default) All network IP settings are provided by the Dynamic Host Control Protocol (DHCP) server in your network. Static – All network IP settings are specified manually through the remaining fields in the LAN section.
IP Address	This setting specifies the IP address that this Interaction SIP Station will use when it starts. This field is used only if the IP Assignment Mode field is set to Static. All IP addresses in managed phones configuration are pass-through strings to support IPv6.
Subnet Mask	This setting specifies the appropriate subnet mask for your IP telephony network. The IP address for this Interaction SIP Station must fall within the range specified by the subnet mask. All IP addresses in managed phones configuration are pass-through strings to support IPv6.
Default Gateway	This setting specifies the IP address for the network device that controls the routing of IP packets. This field is used only if the IP Assignment Mode field is set to Static. All IP addresses in managed phones configuration are pass-through strings to support IPv6.
Primary DNS	This setting specifies the IP address of the primary Domain Name Server (DNS). Do not enter a URL in this field. This field is used only if the IP Assignment Mode field is set to Static. All IP addresses in managed phones configuration are pass-through strings to support IPv6.
Secondary DNS	This setting specifies the IP address of the secondary DNS. Do not enter a URL in this field. This field is used only if the IP Assignment Mode field is set to Static. All IP addresses in managed phones configuration are pass-through strings to support IPV6. Note: The fields for the IP network settings do not reflect the settings that are assigned by the DHCP server to the Interaction SIP Station.

Audio Quality Diagnostics

These settings are designed to help diagnose audio quality problems by capturing recorded audio packets using a network sniffer utility (for example Wireshark) and analyzing the network traffic and audio packets. Change these settings only at the direction of authorized PureConnect Customer Care technical support representatives. Setting these values incorrectly can cause additional audio problems.

Option	Description	Default
Packet Recording Enabled	This setting activates the packet recording mechanism.	No
Packet Recording Remote IP Address	This setting specifies the IP address of the remote computer to capture the recorded packets. The recorded packets should be captured by a network sniffer application. The default IP address is 0.0.0.0. All IP addresses in managed phones configuration are pass-through strings to support IPv6.
Packet Recording Remote Port	This setting specifies the UDP port on the computer specified in Packet Recording Remote IP Address.	5000
RTP Recording Enabled	This setting activates the DSP RTP recording.	No
Network Recording (To User) Enabled	This setting activates the DSP network recording of outgoing audio traffic.	No
TDM Recording (From User) Enabled	This setting activates the DSP TDM recording of incoming audio traffic.	No
Echo Canceller Debug Recording Enabled	This setting activates recording of the echo cancellation activity to debug associated echo problems.	No
Port Mirroring	This setting activates port mirroring.	No
SRTP Audit Events	This setting activates auditing of SRTP events.	No
Noise Reduction Debug Recording	This setting activates noise reduction debug recording.	No

Registration

Option

Description

Default

Allow Manual Redundant Proxy Symmetric

CIC sets the redundant_proxy_is_symmetric parameter to true only when proxy type is configured as Line. In some scenarios, this is problematic and as a result the phone cannot be registered.

Set this option to Yes to allow CIC to set the redundant_proxy_is_symmetric parameter as true if the first and second registration types are manual.

For more information about the use of this option, see the Interaction SIP Proxy Technical Reference.