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Interaction Administrator Help
Advanced options: AudioCodes and Genesys phones or templates
The following tables describe the advanced options for AudioCodes and Genesys 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 setting specifies the Dynamic Host Configuration Protocol (DHCP) option number that identifies the vendor and functionality of a DHCP client. Note: For AudioCodes and Genesys, instead of setting this option, set the DHCP Option Records 002 in the network configuration. For more information, see CIC Managed IP Phones Administrator's Guide in the Technical Reference Documents section of the PureConnect Documentation Library.
|
160 |
Static URL |
This setting specifies 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 AudioCodes or Genesys phone 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> |
Enable Configuration Web Page |
This setting determines whether the phone's configuration web page appears on the AudioCodes or Genesys phones. By default, this configuration page is visible to users. |
On |
Obtain Time Zone from DHCP |
This option enables prioritization of the NTP GMT offset information received from the DHCP server, over the static configuration (system/ntp/gmt_offset). |
No |
Syslog Tracing
The following are all of the Syslog tracing options for AudioCode phones. 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 authorized PureConnect Customer Care representatives. |
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 |
NB Headset Digital Microphone Gain |
This setting controls the gain of the RTP audio stream sent from the AudioCodes or Genesys phone to the line (narrow band). |
+0 dB |
NB Headset Digital Speaker Gain |
This setting controls the gain of the RTP audio stream from the line through the AudioCodes or Genesys phone and to the headset (narrow band). |
+4 dB |
WB Headset Digital Microphone Gain |
This setting controls the gain of the RTP audio stream sent from the AudioCodes or Genesys phone to the line (wide band). |
+0 dB |
WB Headset Digital Speaker Gain |
This setting controls the gain of the RTP audio stream from the line through the AudioCodes or Genesys phone and to the headset (wide band). |
-4dB |
VLAN
Option |
Description |
Default |
VLAN Discovery Mode |
This setting specifies the method that this AudioCodes or Genesys phone 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:
|
|
Manual VLAN ID |
If the VLAN Discovery Mode field is set to "Manual", you can specify the VLAN ID for this AudioCodes or Genesys phone in this field by entering the value it will use. Valid values are 1 through 4094. Note: Important information regarding
Manual VLAN ID
|
|
LAN
Option |
Description |
Default |
IP Assignment Mode |
This setting specifies the method that this AudioCodes or Genesys phone uses to obtain an IP address:
|
|
IP Address |
This setting specifies the IP address that this AudioCodes or Genesys phone 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 AudioCodes or Genesys phone 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 AudioCodes or Genesys phone. |
|
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 option 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 option activates the DSP RTP recording. |
No |
Network Recording (To User) Enabled |
This option activates the DSP network recording of outgoing audio traffic. |
No |
TDM Recording (From User) Enabled |
This option activates the DSP TDM recording of incoming audio traffic. |
No |
Echo Canceller Debug Recording Enabled |
This option activates the recording of the echo cancellation activity to debug associated echo problems. |
No |
Registration
Option |
Description |
Default |
Redundant Proxy Mode |
This option is mandatory for the phone to operate in redundancy. Enable the option to operate with a proxy server that serves as a backup if the first goes down. • Primary-Fallback/Disabled - Phone does not use redundant proxy. If set to Disable in the Web interface, it will set the previously described ini file parameter redundant_proxy/enabled as well. Primary-Fallback -Phone registered to redundant proxy if the primary proxy does not respond to SIP signaling messages. |
Primary-Fallback/Disabled |
Switch Back to Primary SIP Proxy |
The phone identifies cases where the primary proxy does not respond to SIP signaling messages. In these scenarios, the phone registers to the redundant proxy and the phone continues normal functionality, without the user noticing any connectivity failure or malfunction with the primary proxy. No/disabled = Asymmetric (default). In this mode, the primary proxy is assigned a higher priority for registration than the redundant proxy. Once the phone is registered to the primary proxy, it sends keep-alive messages (using SIP OPTIONS messages) to the primary proxy. If the primary proxy does not respond, the phone registers to the redundant proxy, but continues sending keep-alive messages to the primary proxy. If the primary proxy responds to these keep-alive messages, the phone re-registers to the primary proxy. Therefore, the phone assigns the primary proxy a higher priority for registration. If asymmetric mode is configured and the primary server goes down, an attempt will be made to revert to the primary server. Yes/Enabled = Symmetric. In this mode, both proxies are assigned the same priority for registration. Once the phone is registered to a proxy, it sends keep-alive messages to this proxy. The phone switches proxies only once the proxy to whom it has registered does not respond. Therefore, the phone assigns both proxies the same priority for registration. If symmetric mode is configured and the primary server goes down, the phone operates with the redundant proxy without ever reverting to the primary unless the redundant proxy also goes down. In both the Asymmetric and Symmetric modes, the following applies:
|
No |
Failback Retry Timeout |
This options determines whether to perform failback. This option applies only if you are operating with the DNS mode of failover (for example, with a DNS server). Select 0 to disable the failback retry timeout or enter the number of seconds to wait before failback is performed. |
0 |
Enable Keep Alive Using OPTIONS |
This options determines whether keep-alive is performed by using SIP OPTIONS messages sent to the proxy. |
Yes |
Related topics
Configure managed IP phones or templates