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Interaction Administrator Help
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.
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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:
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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 WARNING
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LAN
Option |
Description |
Default |
IP Assignment Mode |
This setting specifies the method that this Interaction SIP Station uses to obtain an IP address:
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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. |
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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. |
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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. |
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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. |
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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. |
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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. |
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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. |
No |
Related topics
Configure managed IP phones or templates