DSX IP Keysets are available on-premise in a managed network or as remote IP extensions.
DSX IP also supports compliant third-party SIP phones, soft phones, and ATAs.
|Setting up DSX Remote IP Keysets using NAT Traversal requires the purchase of a Linksys WRT54GL or WRT160NL router for each remote site. Additionally, port 5060 at the system-side router must be forwarded to the system IP address. Depending on your experience level, this may require the services of an IT professional.|
|If setting up DSX Remote IP Keysets using VPN (Virtual Private Network), it requires the purchase of third-party VPN routers and the services of an IT professional.|
|There are two available DSX IP Keysets: the 34-Button Backlit Display IP Telephone with Full-Duplex Speakerphone and the 34-Button Backlit Super Display IP Telephone with Full-Duplex Speakerphone.|
|34-Button Backlit Display IP Telephone with Full-Duplex Speakerphone (P/N 1090034)|
|The 34-Button Display IP Telephone features a large 3 line- by-24 character backlit alphanumeric display with 4 Interactive Soft Keys for intuitive feature access. It also provides 10 Personal Speed Dial keys, 24 programmable Feature Keys and 12 ﬁxed function keys for streamlined operation. Additionally, this telephone offers a backlit keypad, a headset jack, and built-in Full-Duplex Speakerphone. Unique features include dual LEDs, built-in wall mounting, and an innovative two position angle adjustment.|
|The 34-Button Display IP Telephone has the same features and ease of use as the digital "TDM" model.|
|34-Button Backlit Super Display IP Telephone with Full-Duplex Speakerphone (P/N 1090035)|
|The Super Display IP Telephone features a large 9 line-by-24 character backlit alphanumeric display with 12 Interactive Soft Keys for intuitive feature access. It also provides 10 Personal Speed Dial keys, 24 programmable Feature Keys and 12 ﬁxed function keys for streamlined operation. Additionally, this telephone offers a built-in Full-Duplex Speakerphone, a backlit keypad, and a headset jack. Unique features include dual LEDs, built-in wall mounting, and an innovative two position angle adjustment.|
|The 34-Button Super Display IP Telephone has the same features and ease of use as the digital "TDM" model.|
You can install a DSX IP Keyset on-premise in a managed network or as a remote IP extension. The DSX IP Keyset supports common CODECs (G.711A, G.729, and G.722) which let you easily balance voice quality with available bandwidth.
Setting up DSX Remote IP Keysets using NAT Traversal requires the purchase of a Linksys WRT54GL or WRT160NL router for each remote site. Additionally, port 5060 at the system-side router must be forwarded to the system IP address. Depending on your experience level, this may require the services of an IT professional.
If setting up DSX Remote IP Keysets using VPN (Virtual Private Network), it requires the purchase of third-party VPN routers and the services of an IT professional.
For a complete index of the available VoIP resources, click VoIP Extensions in the menu bar at the top of this page.
The DSX IP Keyset uses a single CAT5 ethernet cable to the desktop which connects to the compact, unobtrusive DSX IP Adaptor. The adaptor connects to the LAN, the IP keyset, and a PC, as well as provide the power required by the telephone. A separate power supply and cable are not required. If the router to which the keyset is connected is Power Over Ethernet (PoE 802.3af) compliant, the DSX IP Adaptor power supply is not required because telephone power can be provided by the router.
|DSX IP Adaptor (P/N 1091045)||
Third party SIP extensions compliant with SIP VoIP RFC 3261 can be installed on-site or remotely. The SIP extension should provide all the features available at a single line telephone unless dial pad keys are "hijacked" by the SIP extension. For example, if a SIP extension uses * as a setup command, it won't be able to use Directed Call Pickup.
When a remote IP extension user dials 911, the call routes to the 911 service for the location in which the main system is installed. The emergency call does not route to the 911 service for the remote location.
|NAT Traversal||Network Address Translation Traversal
|SOHO Router||Small Office Home Office Router
|VPN||Virtual Private Network
[3.00.35] In prior versions, a keyset's Hotline or Call Coverage key BLF was not fully functional when the key was assigned to a third-party SIP extension. For example, there was no BLF indication for any call placed from the third-party SIP extension. Additionally, if a call was ringing the SIP extension, the BLF coverage would continue after the SIP extension user answered. These issues are now corrected.
[3.05] An error is corrected that could cause calls dialed from a 3rd-party SIP telephone to output double-digits on CO lines if the SIP telephone had audio cut-through to the outside line prior to dialing. This could happen, for example, if the user dialed 9, listened for outside dial tone, and then dialed the rest of the number.
This double-digit dialing problem would only occur if the SIP telephone was dialing digits in-band or was using RFC2833 . The system would re-generate digits that were already presented to the outside line in-band.
[3.00.32] The Echo Canceller was inadvertently disabled in version 3.00.30. It is now fully functional.
[3.01] A problem is corrected where swapping two IP extension numbers in programming would cause a one-way audio path..
[3.00.35] The IP address of the internal VOIPDB gateway is changed from 10.0.0.1 to 10.254.254.1. This makes the DSX VoIP more compatible with 10.0.0.xx networks.
[3.01] A correction is made in this version to prevent an IP keyset from becoming stuck in the Connecting to DSX state. The keyset will now automatically de-register so the connection process can begin again.
[3.01] This software version corrects a memory utilization problem that would cause IP performance to continually degrade under heavy traffic. The system would have to be reset to clear the problem.
[3.00.35] Up to 32 IP telephones can be connected to the system, and the number that can be on a call simultaneously is set by the IP port licensing. In prior versions, the licensing also limited the number of IP telephones that could be connected.
[3.00.35] Improvements have been made to the system's IP traffic handling to minimize the likelihood of lockups under heavy IP traffic situations.
[3.00.32] An issue is corrected that could cause IP Keysets to report the MAC address of the VoIPDB after startup or reset. This could potentially cause no audio.
[3.00.32] If an IP keyset is replaced at the same extension, the MAC address reported to the system matches the new keyset. In prior versions, the old address would be retained until reboot potentially causing no audio.
[3.00.32] This option is added in this release but is currently not used.
[3.01] A problem is corrected where IP telephones would get one-way audio if DHCP was enabled and the system was reset.
[3.00.35] In prior releases, pressing SPEAKER while the IP keyset displays Connecting will cause the keyset to continually reset. This is corrected.
[3.00.32] The RTP port range is the setting of 1801-04: RTP Range Start plus 64. This accommodates the 32 RTP ports and 32 RTCP ports required by the system's 32 maximum VoIP ports. In prior versions, the upper end of the RTP range was set in 1801-05: RTP Range End and was 65536 by default.
[3.05] A problem is corrected which would cause the system's VoIP processes to crash if the G.723 Codec was selected in 1812: Codec Type for the active VoIP Profile. Note that G.723 only is only supported by some 3rd-party SIP telephones. It is not a valid Codec type for DSX IP keysets.
[3.05] In prior version 3 releases, a 3rd-party SIP telephone could not negotiate past the first Codec in the 1812: Codec Type list. This is corrected.
[3.00.32] The telephone programming interface for 1812: Codecs is improved. After selecting a profile, pressing Hold now moves through the options for the selected profile.
[3.05] The system forces third-party SIP telephones to reregister every 180 seconds. This minimizes the possibility that third-party SIP telephones will deregister and go out of service.