GigasetN870IP (SDG5)

Description

This resource allows to connect compatible handsets to the Gigaset N670/N870 IP Pro base station in DECT Manager mode.

Release notes

Version 2.1.1 - Early deployment
  • Improvement: Transport must be configurable (PC-3557)
  • Dependency:
    • Gigaset Phone Support >= 1.7.0
    • DHCP Server >= 2.5.0 (if used)
    • SMP >= 5.17.0

Version 2.1.0 - Early deployment
  • Improvement: use new Line ID field for ipui instead of dedicated var (PC-3088)
  • Bugfix: Syntax issue in gen_sip.conf prevented outgoing calls (PC-3279)
  • Bugfix: Don't set OutgoingDialPrefix in order to accept E.164 format for called party (PC-3279)
  • Dependency:
    • Gigaset Phone Support >= 1.7.0
    • DHCP Server >= 2.5.0 (if used)
    • SMP >= 5.17.0

Version 2.0.1 - Early deployment
  • Bugfix: Extension was not set as caller ID (PC-3669)
  • Dependency:
    • Gigaset Phone Support >= 1.6.0
    • DHCP Server >= 2.5.0 (if used)
    • SMP >= 5.15.0

Version 2.0.0 - Early deployment
  • Feature: full sync_code rewriting for source SOP configuration and FMA support (PC-3010)
  • Dependency:
    • Gigaset Phone Support >= 1.6.0
    • DHCP Server >= 2.5.0 (if used)
    • SMP >= 5.15.0

Version 1.1.1 - Early deployment
  • Bugfix: Incorrect CallerID (TS-10557)
  • Dependency:
    • Gigaset Phone Support >= 1.4.0
    • DHCP Server >= 2.5.0 (if used)
    • ALERT! When upgrading from version <= 1.0.1, please set following configuration in the resource
    • "Use TLS" to "yes"
    • "Use SRTP" to "yes"
    • "RTP range ID" to "1"

Version 1.1.0 - Early deployment
  • Improvement: Potential update impact level 1 DONE: no critical impact expected. Update can be applied without risk of breaking critical functionality. Transport must be configurable (PC-3557)
  • Dependency:
    • Gigaset Phone Support >= 1.4.0
    • DHCP Server >= 2.5.0 (if used)
    • ALERT! When upgrading from version <= 1.0.1, please set following configuration in the resource
    • "Use TLS" to "yes"
    • "Use SRTP" to "yes"
    • "RTP range ID" to "1"

Version 1.0.1 - Early deployment
  • Bugfix: no more < > on caller id to prevent breaking XML config (PC-3010)
  • Dependency:
    • Gigaset Phone Support >= 1.4.0
    • DHCP Server >= 2.5.0 (if used)

Version 1.0.0 - Early deployment
  • Feature: Initial Release (M23222)
  • Dependency:
    • Gigaset Phone Support >= 1.4.0
    • DHCP Server >= 2.5.0 (if used)

Resource configuration interface

GUI unavailable.

This resource stands for a handset to connect to a Gigaset N670/N870 base station in DECT Manager mode.
However, the resource doesn't only generate the configuration for the handset, but a more global file able to provision the DECT Manager with general configuration like the VoIP Provider profile (see here).

A handset operates at the DECT level and therefore doesn't have any MAC address but an analogous 'IPUI' (see below). Therefore, in the MAC field, you have to enter the MAC address of the DECT Manager, so the link between the handset and the device that will manage it can be established.

Internal doc about Gigaset devices is here
You can also find detailed stuff in Gigaset's wiki here

NOTE : From SDG5 version 2.1.0, we get rid of the IPUI parameter and use instead the new 'Line ID' field. Thus a headset resource is uniquely identified by the pair (MAC=MAC of DM, Line ID=IPUI of handset).

Resource configuration parameters

  • [ Line ID ] : from 2.1.0, used instead of IPUI parameter below, with the same meaning.

  • Password: password used during the SIP authentication towards the provisioning server.
  • DTMF mode:
    • inband: DTMF signalling through the same channel as voice.
    • rfc2833: DTMF signalling using RTP packets, distinct from the audio packets.
  • Use Remote Party ID header:
    • Default: Keep default settings as defined in the "Communication Server" module configuration.
    • Yes: Allow the server to perform the re-invite procedure.
    • No: Do not allow the server to perform the re-invite procedure.
  • SIP server feature: Note: These technical settings if improperly configured can have impact on the availability of the phone
    • SIP re-invite: allow the voice media to be transferred directly between SIP endpoints through the SIP re-invite procedure. Possible value are:
      • Default: Keep default settings as defined in the "Communication Server" module configuration.
      • Yes: Allow the server to perform the re-invite procedure.
      • No: Do not allow the server to perform the re-invite procedure.
    • Qualify SIP devices registration: In order to monitor the SIP device availability, the Telephony server can send SIP OPTIONS messages (every 60s). If the device does not respond within the defined delay, it will be considered as off-line. Possible values are:
      • Default: Keep the default setting defined in the "Communication Server" module configuration.
      • No: Disable Qualify SIP registration for this device
      • Yes: Enable Qualify SIP registration for this device, with a default timeout value of 2000ms
      • Suggested number: Enable Qualify SIP registration for this device, with the chosen timeout value (expressed in milliseconds).
    • Force specific codec: If this parameter is set to 'Yes', the SIP server will only accept the 3 codecs defined in this resource with the specified preference.
    • First Codec, Second Codec, Third Codec: Ordered list of accepted codecs. ALERT! You should set at least one of those 3 codecs in order for the handsets to work properly!
    • IPUI: Here you have to specify the handset ID (called IPUI (International Portable User Identity)) of the handset you're trying to connect (this is not a MAC address and thus differs from the base station address that you have to specify in the MAC Address field).
      Find it on the device's box or by menu (Idle screen ► menu button ► key combination ✱#06# ► first line)
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