DSX System Software Version 3.2 is an important new release that features three major VoIP enhancements: Peer-to-Peer communication between IP extensions, NAT Traversal with Any Router, and Improved ATA and SIP SLT Support.
DSX System Software Version 3.2 also provides the new IntraMail Message Review feature, enhanced Cascading Message Notification with IntraMail Pro, and improved maintenance functions when used with System Administrator 3.1 or higher.
Peer-to-Peer (p2p)
IP keysets, ATAs, and SIP SLTs connected to another SIP device in the same office do not use VoIP Gateway circuits. They communicate p2p instead. This means you can have 32 SIP devices in an office talking to each other and never use up a VoIP License.
A VoIP License is still required, however, when a SIP device is:
•On an outside call.
•Participating in a Conference.
•Using Room Monitor.
•Using an IntraMail mailbox or leaving a message.
•Calling a SIP device in another office. For example, if an on-premise IP keyset calls an IP keyset in a remote office, each IP phone requires a VoIP License.
By default, Peer-to-Peer is enabled for all IP extensions. The system will automatically switch VoIP gateways in and out as required by the call state.
Note that some 3rd party ATAs and SIP SLTs may not support p2p in the remote office. However, the Cisco PAP2T and SPA8000 do support p2p in the remote office.
NAT Traversal With Any Router for DSX IP Keysets
NAT Traversal is available with any router without programming. You no longer have to limit your remote offices to the two LinkSys routers we previously recommended. To set up NAT Traversal:
•On the phone system router, forward UDP ports 5060 (SIP) and 1024-1088 (RTP) to the DSX system IP address.
•On the DSX IP Keysets, use IP Keyset firmware 1.0.7.2 or higher.
Improved ATA (Analog Telephone Adapter) and SIP SLT Support
Major improvements have been made to the DSX integration with third-party ATAs and SLTs. In general, ATAs and SIP SLTs work the same as local analog single line sets.
Internal Modem Automatically Disabled when VoIP Daughter Board Installed
To protect the viability of VoIP resources when the VoIP Daughter Board is installed, the internal modem is automatically disabled. The reliability of DSX VoIP could be compromised if both were enabled simultaneously.
Hot Dial Pad Follows Prime Line
If an extension has Hot Dial Pad enabled 2111-11:Hot Dial Pad and a Prime Line Set in 2114-01: Prime Line, any digits the user dials on the dial pad will go out over the Prime Line. In prior versions, Hot Dial Pad would always follow Intercom Prime Line.
Hotline Keys for Voice Mail Ports
An extension user can assign IntraMail ports (501-508) to Hotline keys in Menu: Key Assignment: Feature Keys (Menu + 51). This can be a useful visual tool for monitoring voice mail port usage.
Important New IntraMail Features
Message Review
After a caller leaves a message they can review their message before sending. After reviewing the message they can rerecord it, erase it, or send it to the recipient unchanged. Message Review is available to both co-workers and outside callers.
Cascading Message Notification Enhanced
For added flexibility, IntraMail Pro Cascading Message Notification now allows multiple cycles through the list of destinations. You can, for example, have notification cascade through each destination once and then repeat the cycle. Additionally, the number of busy and RNA attempts per destination can be from 1-99, and the total number of notification cycles can also be from 1-99. This allows for very complex notification cascades if the site requires.
The meaning of the following timers have changed to accommodate these enhancements.
•4202-06: Maximum Attempts sets how many times Message Notification cascades through the list of destinations.
•4202-01: Notification Interval sets how long IntraMail waits, after completing the cascade, before starting again. In the prior release, a new cascade would start immediately after the prior cascade finished with no delay in between.
IntraMail Maintenance and Prompt Update Features
System Software 3.2 enables all the System Administrator Version 3.1 Maintenance and Prompt Update features. This includes:
•Copying, moving, and erasing mailboxes.
•Deleting voice prompt sets that are already installed (e.g., to make more message storage room).
Automated Attendant Call Progress Tone Detection
If an outside line without loop disconnect supervision is answered by the Automated Attendant, and caller hangs up, Call Progress Tone Detection will detect the busy, reorder, or dial tone on the line and hang up. This prevents the call from timing out to the Dial Action Table Timeout destination and, in some configurations, inadvertently setting up a loop.
Additional Silence Elimination for Messages
IntraMail will automatically remove 4.5 seconds of silence from any message recording that is paused due to silence. In addition to cleaning up normal messages, it will also help to eliminate "hang up" messages because they will be too short to be saved.
Software Corrections
•3rd Party SIP Cannot Call Ring Group, ICM Queue Key, or Busy UCD Group
Software Version 3.20 allows 3rd Party SIP SLTs and ATAs to call Ring Groups, busy extensions with an ICM Queue key, and busy UCD Groups. These operations were not available in prior 3.1x releases.
•ATA Handling of Hold, Transfer, and Park Improved
The handling of Hold, Transfer, and Park at third-party ATAs is improved.
◦Hold
- To put a call on Exclusive Hold at an ATA: Hookflash + Dial *7# + Hang up.
- To pick up the call on Exclusive Hold: Lift handset + Dial *7#.
◦Transfer
- To Transfer a call: Hookflash + Dial extension + # + Hang up.
- To retrieve the call if you co-worker doesn't want it: Hookflash + *7#.
◦System Park
- To Park a call in System Orbit: Hookflash + * + Orbit (60-69)
- To retrieve a call from System Orbit: Lift handset + * + Orbit (60-69)
◦Personal Park
- Personal Park is not available because ** is not allowed by the ATA.
•Barge-In Beeps on IP Telephones
Barge-In beeps will now correctly occur when an IP telephone is using or receiving a Barge In.
•Caller ID Display Enhancements for 3rd Party SIP Telephones
The Caller ID display information provided to 3rd Party SIP Telephones and ATAs is enhanced for more consistency with analog single line telephones.
•Cascading Message Notification Enhanced
The meaning of the following timer has changed to further improve Enhanced Message Notification.
◦4202-01: Notification Interval sets how long IntraMail waits, after completing the cascade, before starting again. In the prior release, a new cascade would start immediately after the prior cascade finished with no delay in between.
•Conference and ATAs
Many improvements were made to the handling of third-party ATAs when included in a Conference.
•DHCP Support
The execution of the DHCP with Manual IP option is improved.
•Dial Preview Without Dialing
A potential system reset is corrected that could occur when a user dialed ** and then lifted the handset without dialing any digits.
•Disabling Multicast Paging Over the LAN
The option 2106-05: LAN Multicast provides the option to disable Multicast Paging over the local LAN. If a local IP keyset is connected to an Ethernet switch that blocks multicast, it will only broadcast Paging if you disable Multicast Paging for that extension.
Note that if you disable this option, every IP phone receiving paging will require a unique VoIP gateway channel. Multicast uses one VoIP gateway channel regardless of the number of phones being paged.
•Distribution Lists Re-enabled
Software version 3.20 re-enables the Distribution List feature. This feature was inadvertently disabled in software version 3.11.
•DSX-80/160 DSP Version 3.13
Corrects a potential backplane communication problem affecting interface
card firmware upgrades when VOIP DB is installed.
•Flexible TCP Port Assignment
The option 1104-06: TCP Port is added to enable the System Administrator Flexible TCP Port Assignment feature.
•Improvements to VoIP Peer-to-Peer Connections
Additional enhancements to VoIP P2P connections are provided in this release to improve operation and maximize gateway resources.
•IntraMail Message Notification Over PRI DID
This software version corrects a system reset that would occur when a PRI DID line was used for outbound IntraMail Message Notification. This would occur only when the previous incoming call had an un-translated DID number.
•No Speech Path At IP Keyset When Speed Dial Uses Line Group
In prior 3.1x software versions, an IP keyset would have no speech path after selecting a Speed Dial bin that used a line group. This is fixed in version 3.20.
•No Speech Path from Prime Line
Software Version 3.20 corrects an issue inadvertently introduced by 3.17 in which VoIP keyset users would have no speech path from their Prime Line (Line or Line Group).
•Peer-to-Peer Across the WAN
The option 2106-04: P2P over WAN is an advanced option for setting up peer-to-peer connections between two SIP devices at different remote offices or between a local and a remote SIP device. This is an option to be used by advanced users with IT experience only, as it requires knowledge and understanding of network address translation and RTP protocols. Additional information will be provided.
•Peer-to-Peer Performance With PAP2T ATA Improved
In software version 3.20, doing a Make Call (MC) between two SLTs connected to the same PAP2T ATA no longer requires special setup. In prior software versions, speech path problems would likely result and required port forwarding the PAP2T RTP ports or disabling P2P on the LAN.
•PRI Field Problem When Using Forced Account Codes Corrected
Software version 3.20 corrects a field problem that could cause a system reset in a busy system with Forced Account Codes enabled.
•PRI Firmware
The PRI Card firmware is updated to 4.6 to correct an issue found with specific network providers that would result in an immediate disconnect of calls to busy destinations.
•PRI Firmware Corrects Nortel NI-2 Field Issue
T1/PRI Card firmware is updated to 4.6 to correct a specific field-reported Nortel NI-2 Busy issue.
•Programmable SIP Registration Expiry Time
The SIP Registration Expiry Time (the interval after which a SIP device's registration expires) is now programmable using option 1802-02: Reg. Expiration. The default interval (40 seconds) is normally adequate.
In DSX, the SIP Registration packets serve a dual purpose: they keep the IP device registered with the system and, for remote IP phones and other SIP devices, they function as SIP “keep alive” messages. This ensures that IP port translations remain open in the NAT router and that the connection to the DSX remains active. If SIP traffic is not sent often enough, the NAT router may close the IP port and cause the remote phone or device to be disconnected. If you find your SIP device continually connecting and disconnecting from the DSX you may need to decrease the SIP Registration Expiry Time.
•Ring No Answer Between Two ATAs Could Cause IP Lockup
In prior 3.1x software versions, an unanswered Intercom call between two SLTs connected to the same ATA could result in a lockup of the IP processes. This is fixed in software version 3.20.
•User Programmable Features Access Levels Match
The access levels for the Menu key user features now match the access levels for the legacy on-hook #xx codes. In addition, the access levels for some options have been changed to provide more feature differentiation among the different levels.