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Switch |
This menu is accessed by selecting System in the menu bar and clicking on Switch.
The following settings are shown in this panel:
• | System Name A name used to identify the system. This is typically used to identify the configuration by the location or customer's company name. Some features require the system to have a name. This field is case sensitive. Do not use <, >, |, \0, :, *, ?, . or /. |
• | Mode: The system can operate in either Key or PBX mode. Changing the mode requires the IP Office system to be restarted and will overwrite all existing button programming. |
• | Key The Number of Lines setting is used to automatically assign line appearance buttons on all extensions with programmable buttons. To make external calls the user should select an available line appearance button. Outbound call routing is determined by which line appearance button the user selects before dialing or by the user's automatic line selection settings. |
• | PBX No line appearances are automatically assigned to programmable buttons. The Outside Line setting is used to set the dialing prefix that indicates that the call is an external one for which an available line should be seized. The Alternate Route Selection settings are used to determine which lines are used for each outgoing call. Line appearance buttons can also still be configured for making and answering external calls. |
• | Voicemail Mode: Default = Intuity Mode. Software level = 8.0+. Embedded voicemail can use either IP Office Mode or Intuity Mode key presses for mailbox functions. End users should be provided with the appropriate mailbox user guide for the mode selected. Pre-Release 8.0 systems use IP Office Mode only. |
• | IP Address This field sets the address of the PC allowed to send files to the system's memory card. |
• | Country: This option sets a range of country specific telephony settings. The system language can be changed from the Country setting using the separate Language setting below. |
• | ! WARNING - Reboot Required Changing this setting requires the system to be rebooted for the change to take effect. Rebooting the system will end all calls currently in progress. |
• | The supported countries are Argentina, Australia, Bahrain, Belgium, Brazil, Canada, Chile, China, Customize, Denmark, Egypt, Finland, France, Germany, Greece, Hong Kong, Hungary, Iceland, India, Italy, Korea, Kuwait, Mexico, Netherlands, New Zealand, Norway, Oman, Pakistan, Peru, Poland, Portugal, Qatar, Russia, Saudi Arabia, Singapore, South Africa, Spain, Sweden, Switzerland, Taiwan, Turkey, United Arab Emirates, United States, Venezuela. |
• | When Default is selected, the following additional fields are available: |
• | Tone Plan: Default = Tone Plan 1 Select a tone plan to be used for different ringing signals such as dial tone and ring tone. |
• | CLI Type: Default = FSK V23 Set the method for passing caller ID information to analog extensions. The options are DTMF, FSK Bell 202 or FSK V23. |
• | Busy Tone Detection: Default = Off Enable or disable the use of busy tone detection for call clearing. |
• | Language This field sets the language used for voicemail prompts and phone displays if the language is available. Possible languages are: |
• | Arabic, Brazilian Portuguese, Canadian French, Cantonese, Danish, Dutch, Finnish, French, German, Italian, Korean, Mandarin, Norwegian, Portuguese, Russian, Spanish, Spanish (Argentinean), Spanish (Latin), Spanish (Mexican), Swedish, UK English, US English. |
• | ! WARNING - Reboot Required Changing this setting requires the system to be rebooted for the change to take effect. Rebooting the system will end all calls currently in progress. |
• | For each user, their language settings can be changed using the user's Language setting. This affects the language used on their phone's display and for mailbox access prompts. |
• | For each auto attendant, the system language setting can be overridden by the auto attendant's own Language setting. |
• | Password: Default = Blank. Range = 4 digits. This is a four digit code used to restrict access to some functions. Once set, the system password must be used to override station lock, forced account, numbers in the disallowed calls list or night service restrictions to make a call. The system password is also requested when a user switches the phone system into or out of night service mode or tries to access an voicemail auto attendant's emergency greeting settings. |
• | For M-Series and T-Series phones, the system password, if set, is also used to control access to phone based administration from the first two extensions in the system. |
• | Number of Lines: Default = The number of analog trunks present when the system is first started. This option is only available for systems with their Mode (see above) set to Key. For phones with programmable buttons, those buttons can be configured as line appearance buttons that each match a particular incoming line. This setting controls how many of buttons on every user's phone are automatically allocated as line appearance buttons. The assignment is done starting from button 03 upwards in order of the lines available. |
• | ! Warning If the Number of Lines value is changed, all existing line appearance buttons and automatic line selection settings are overwritten. The existing functions on other programmable buttons are also overwritten if they are in the range of buttons now specified for lines. Therefore it is recommended that this setting is only changed when a system is first installed. |
• | Outside Line: Default = Depend on system locale, see below. This option is only available for systems with their Mode (see above) set to PBX. It sets the digit which, when dialed, indicates that the call is intended to be external. Routing of any additional digits is then determined through the Alternate Route Selection settings. |
• | ! WARNING - Reboot Required Changing this setting requires the system to be rebooted for the change to take effect. Rebooting the system will end all calls currently in progress. |
• | 9 (Operator is 0) The prefix 9 is used for external calls. The digit 0 is used for calls to the operator extension (the first extension in the system). This is the default setting for systems with the Country setting United States. |
• | None No prefix is used for external calls. Any dialing that does not match an internal dial plan number is assumed to be an external call. This is the default setting for systems with the Country setting other then Germany or United States. The digit 0 is used for calls to the operator extension (the first extension in the system). |
• | 0 (Operator is 9) The prefix 0 is used for external calls. The digit 9 is used for calls to the operator extension (the first extension in the system). This is the default setting for systems with the Country setting Germany. |
• | Log All Caller ID Calls for Users: Default = None selected. All extensions have a call log of their last 30 calls (incoming answered and missed). The user can access this using a programmable button set to Call Log or their phone's Call Log or History button if it has one. In addition, up to 3 extensions can be configured to have access to the call log of the last 400 calls (incoming answered and missed) for the whole system. These fields are used to select those users. Only calls that include caller ID are included. The ! character on the phone display indicates that there are unviewed call details in the call log. |
• | Allow Unsupervised Analog Trunk Disconnect: Default = No. When using analog trunks, various methods are used for trunk supervision, ie. to detect when the far end of the trunk has disconnected and so disconnect the local end of the call. Depending on the locale, the system uses Disconnect Clear signalling and or Busy Tone Detection. This setting should only be enabled if it is known that the analog trunks do not provide disconnect clear signalling or reliable busy tone. When enabled: |
• | Disconnect clear signalling detection is turned off. Busy tone detection remains on. |
• | Unsupervised transfers and trunk-to-trunk transfers of analog trunk calls are not allowed. |
• | A wider range of busy tones which may signal that the caller has disconnected are used to disconnect calls connected to voicemail. |
• | When this setting is changed to No, the configuration settings for Busy Tone Detection are displayed. |
• | Mode: Default = System Frequency If set to System Frequency, the settings used are the default settings for the system locales. To change the settings, select either Single Frequency or Dual Frequency to match the line providers requirements. |
• | Single Frequency If the Mode is set to Single Frequency, set the frequency. |
• | Dual Frequency If the Mode is set to Dual Frequency, set the frequencies. |
The following settings are shown in this panel.
• | ! WARNING - Reboot Required Changing any of these settings requires the system to be rebooted for the changes to take effect. Rebooting the system will end all calls currently in progress. |
• | Receive IP Address Via DHCP Server: Default = Yes. This setting controls whether the system acts as a DHCP client or uses a fixed IP address. |
• | If enabled, the system acts as a DHCP client and requests IP address details for its LAN port when the system is started. |
• | If it receives a response, the address details it has been given by the DHCP server are shown in the field below but cannot be adjusted. |
• | If it does not receive a response, it default to using the address 192.168.42.1. It is still a DHCP client and will request an address again when it is next restarted. |
• | If not enabled, the system uses the IP address values set in the fields below. |
• | System IP Address: Default = 192.168.42.1 Enter the IP address that the telephone system should use if Receive IP Address Via DHCP Server is not selected. If Receive IP Address Via DHCP Server is selected, this field is greyed out but does display the IP address that the system is currently using. |
• | Subnet Mask: Default = 255.255.255.0 Enter the Sub-Net Mask that the telephone system should use if Receive IP Address Via DHCP Server is not selected. If Receive IP Address Via DHCP Server is selected, this field is greyed out but does display the IP address that the system is currently using. |
• | Default Gateway: Default = 0.0.0.0 Enter the Default Gateway that the telephone system should use if Receive IP Address Via DHCP Server is not selected. If Receive IP Address Via DHCP Server is selected, this field is greyed out but does display the IP address that the system is currently using. |
• | DNS Settings This option is greyed out on systems configured to use DHCP as the DHCP server will provide DNS information. For systems not using DHCP, clicking DNS Settings displays the DNS Settings menu. |
• | DNS Server IP Address This field sets the address for the primary DNS server that the system should use to try to resolve domain names to IP addresses. |
• | Backup DNS Server IP Address This field sets the address for the secondary DNS server that the system should use if there is no response from the primary DNS server. |
The following settings are shown in this panel:
• | Companding Law The system is automatically defaulted to A-Law or U-Law by the type of SD Feature Key dongle inserted into the unit. Typically U-Law is used in North American locales, A-Law is used in most other locales. U-Law is also called Mu-Law or µ-Law. For some installations, it may be necessary to change this setting if advised by the external line provider. |
• | ETR6 cards are not supported for systems running in A-Law mode. |
• | Automatic DST: Default = On. When selected, the telephone system will automatically apply daylight saving time (DST) adjustments to its internal clock. This feature should only be used for systems in a North American locale. |
• | Enable Network Time Synchronization: Default = On. When selected, the system will use the time included in the ICLID on incoming calls as its system time. Note that this feature uses the first analog trunk on the card installed in slot 1 of the system control unit. |
This menu is accessed by clicking the Advanced button on the System menu.
Advanced System Parameters
The following settings are shown in this panel:
• | Ring on Transfer: Default = Active. If selected, callers being transferred hear ringing during the transfer process. If not selected, the caller will hear music on hold. |
• | Hold Reminder Time: Default = 60 seconds. Range = 0 (Off) to 180 seconds. This setting controls how long calls remain on hold before recalling to the user who held the call. Note that the recall only occurs if the user has no other connected call. Recalled calls will continue ringing and do not follow forwards or go to voicemail. |
• | Recall Timer Duration: Default = 500. Range = 25 to 800 milliseconds. This is the flash pulse width used for analog trunks and T1 trunks. |
• | ! WARNING - Reboot Required Changing this setting requires the system to be rebooted for the change to take effect. Rebooting the system will end all calls currently in progress. |
• | Transfer Return Ring: Default = 4 (20 seconds), Range 1 to 180 seconds. Sets the delay after which any call transferred by a user that remains unanswered, should return to the user. A return call will continue ringing and does not follow any forwards or go to voicemail. Transfer return will occur if the user has an available call appearance button. Transfer return is not applied if the transfer is to a hunt group. |
• | Outside Conference Denial: Default = Allowed. When set to the Allowed, more than one outside line can be added to a conference. When set to the Disallowed, a second outside line can not be added to a conference. This feature does not change based on the type of outside line. The intent of this feature is to minimize toll fraud. For example, if set to disallowed, this would prevent someone from accepting an outside call at an extension, conferencing in another outside party, and then walking away allowing the two parties to converse. |
• | Toll Call Prefix: Default = 0 or 1 Required Before Area Code. Allows selection between 0 or 1 Required Before Area Code or Area Code and Number Only. |
• | ! WARNING - Reboot Required Changing this setting requires the system to be rebooted for the change to take effect. Rebooting the system will end all calls currently in progress. |
• | Default Name Priority: Default = Favour Trunk. Software level = 8.0+. 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 system speed dials. This setting determines which method is used by default. For each SIP line, this setting can be overridden by the line's own Name Priority setting if required. |
• | 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. |
• | Favour Directory Search for a number match in the system speed dials. 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 is used. |
STUN Settings for Network
These settings are used if SIP trunks are added to the phone system's configuration using the SIP Trunk Administration menu. These settings are necessary to allow SIP connections from the network on which the phone system is attached to reach the public network on which the SIP provider is located.
The following fields can be completed either manually or the phone system can attempt to automatically discover the appropriate values. To complete the fields automatically, only the STUN Server IP Address is required. STUN operation is then tested by clicking Run STUN. If successful the remaining fields are filled with the results.
• | ! WARNING - Reboot Required Changing any of these settings requires the system to be rebooted for the changes to take effect. Rebooting the system will end all calls currently in progress. |
• | Enable STUN: Default = Off This field is used to select whether STUN is used or not. |
• | STUN Server IP Address: Default = Blank This is the IP address of the line providers SIP STUN server. The phone system will send basic SIP messages to this destination and from data inserted into the replies can try to determine the type ITSP NAT changes being applied by any firewall between it and the ITSP. |
• | STUN Port: Default = 3478 Defines the port to which STUN requests are sent if STUN is used. |
• | Firewall/NAT Type: Default = Unknown The settings here reflect different types of network firewalls. |
• | Blocking Firewall Allow outgoing TFTP WRQ. Typically this will be the case. It has been observed that the Avaya corporate firewall permits outgoing TFTP RRQ. |
• | Symmetric Firewall SIP packets are unchanged but ports need to be opened and kept open with keep-alives. If this type of NAT is detected or manually selected, a warning ‘Communication is not possible unless the STUN server is supported on same IP address as the ITSP will be displayed as part of the manager validation. |
• | Open Internet No action required. If this mode is selected, STUN lookups are not performed. |
• | Symmetric NAT A symmetric NAT is one where all requests from the same internal IP address and port, to a specific destination IP address and port, are mapped to the same external IP address and port. If the same host sends a packet with the same source address and port, but to a different destination, a different mapping is used. Furthermore, only the external host that receives a packet can send a UDP packet back to the internal host. SIP Packets need to be mapped but STUN will not provide the correct information unless the IP address on the STUN server is the same as the ITSP Host. If this type of NAT/Firewall is detected or manually selected, a warning ‘Communication is not possible unless the STUN server is supported on same IP address as the ITSP’ will be displayed as part of the manager validation. |
• | Full Cone NAT A full cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Furthermore, any external host can send a packet to the internal host, by sending a packet to the mapped external address. SIP packets need to be mapped to NAT address and Port; any Host in the internet can call in on the open port, that is the local info in the SDP will apply to multiple ITSP Hosts. |
• | Restricted Cone NAT A restricted cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Unlike a full cone NAT, an external host (with IP address X) can send a packet to the internal host only if the internal host had previously sent a packet to IP address X. SIP packets needs to be mapped. Responses from hosts are restricted to those that a packet has been sent to. So if multiple ITSP hosts are to be supported, a keep alive will need to be sent to each host. If this type of NAT/Firewall is detected or manually selected, no warning will be displayed for this type of NAT. |
• | Port Restricted Cone NAT A port restricted cone NAT is like a restricted cone NAT, but the restriction includes port numbers. Specifically, an external host can send a packet, with source IP address X and source port P, to the internal host only if the internal host had previously sent a packet to IP address X and port P. SIP packets needs to be mapped. Keep-alives must be sent to all ports that will be the source of a packet for each ITSP host IP address. If this type of NAT/Firewall is detected or manually selected, no warning will be displayed for this type of NAT. However, some Port Restricted have been found to be more symmetric in behavior, creating a separate binding for each opened Port, if this is the case the manager will display NATs a warning ‘Communication is not possible unless the STUN server is supported on same IP address as the ITSP’ as part of the manager validation. |
• | Unknown Use this setting if the other settings are unsuitable |
• | Static Port Block Use the RTP port range 49152 to 53246. |
• | Binding Refresh Time (seconds): Default = 0 (Never). Range = 0 to 3600 seconds. Having established which TCP/UDP port number to use, either through automatic or manual configuration, the phone system can send recurring ‘SIP Options requests’ to the remote proxy terminating the trunk. Those requests will keep the port open through the firewall. Requests are sent every x seconds as configured by this field. If a binding refresh time has not been set you may experience problems receiving inbound SIP calls as they are unable to get through the Firewall. In these circumstances make sure that this value has been configured. |
• | Public IP Address: Default = 0.0.0.0 This value is either entered manually or discovered by the Run STUN process. If no address is set, the phone system IP address is used. |
• | Public Port: Default = 0 This value is either entered manually or discovered by the Run STUN process. |
• | Run STUN This button tests STUN operation between the phone system and the STUN Server IP Address set above. If successful the results are used to automatically fill the remaining fields with the discovered values. Before using Run STUN the SIP trunk must be configured. |
SMTP Server Configuration
Email can be used to provide users with an alert when they have a new voicemail message. This feature is called voicemail email. This requires the system to be configured with details of an SMTP email server account which is used to forward the messages to the user's email address.
• | ! WARNING - Reboot Required Changing any of these settings requires the system to be rebooted for the changes to take effect. Rebooting the system will end all calls currently in progress. |
• | IP Address: Default = 0.0.0.0 This field sets the IP address of the SMTP server being used to forward emails. |
• | Port: Default = 25. Range = 0 to 65534. This field sets the destination port on the SMTP server. |
• | Send Email From: Default = Blank This field sets the sender address to be used for emails from the system. Depending of the authentication requirements of the SMTP server, this may need to be a valid email address hosted by that server. Otherwise the SMTP email server may need to be configured to allow SMTP relay of this address. |
• | Server Authentication: Default = On This field should be selected if the SMTP server being used requires authentication to allow the sending of emails. When selected, the User Name and Password fields become available. |
• | User Name: Default = Blank This field sets the user name to be used for SMTP server authentication. |
• | Password: Default = Blank This field sets the password to be used for SMTP server authentication |
• | Enable CRAM-MD5: Default = Off. This field should be selected if the SMTP server uses CRAM-MD5. |
© 2013 AVAYA - Issue 02..01 9:42 AM, Wednesday, July 31, 2013 (system2.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 other protective rights of Avaya or others. All trademarks identified by (R) or TM are registered trademarks or trademarks respectively of Avaya Inc. All other trademarks are the property of their respective owners. Last Modified: 4/8/2013 |