Cisco SPA8000 (SDS6)

Description

This resource allows to connect a Cisco SPA8000 analogue telephony adapter.

Release notes

Version 2.0.0 - Early deployment
  • Feature: full sync_code rewriting for source SOP configuration and FMA support (PC-3925)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+
    • Communication Server module 2.0.0+ (If T.38 support is required)

Version 1.10.0 - Early deployment
  • Feature: Support Calls over Internet (M16268)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+
    • Communication Server module 2.0.0+ (If T.38 support is required)

Version 1.9.2 - Early deployment
  • Bugfix: ToS value proposed on the dropdown were incorrect (M21105)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+
    • Communication Server module 2.0.0+ (If T.38 support is required)

Version 1.9.1 - Early deployment
  • Bugfix: Allow space in dialplan (M20272)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+
    • Communication Server module 2.0.0+ (If T.38 support is required)

Version 1.9.0 - General deployment
  • Feature: Allow the '*' character in the dialplan (M0)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+
    • Communication Server module 2.0.0+ (If T.38 support is required)

Version 1.8.0 - General deployment
  • Feature: Add codec selection for fax passthru (M8788)
  • Bugfix: Using the preferred codec could change the codec preference on other resources (M9787)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+
    • Communication Server module 2.0.0+ (If T.38 support is required)

Version 1.7.1 - Deprecated
  • Bugfix: When you dialed internal numbers like 1000, the belgian dialplan was calling emergency numbers (M6693)
  • Deprecated: In certain cases, using this resource can negatively impact the codec negotiation of SIP trunks or SIP devices (M9787)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+
    • Communication Server module 2.0.0+ (If T.38 support is required)

Version 1.7.0 - Deprecated
  • Potential update impact level 2 DONE: in the event this update contains a bug, it might have critical impact. Respect dependencies and retest your most important callflows and applicative integrations.: The default dialplan has changed from a US numbering plan to a Belgian numbering plan. Upgrading to this version should be tested before applying it on all devices.
  • Feature: Configurable dialplan to allow faster dialing after the number has been completed (M6063)
  • Deprecated: In certain cases, using this resource can negatively impact the codec negotiation of SIP trunks or SIP devices (M9787)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+
    • Communication Server module 2.0.0+ (If T.38 support is required)

Version 1.6.1 - Deprecated
  • Bugfix: Receiving T38 faxes now also works (M0005958)
  • Improvement: Removed unsupported codecs from the codec selection list
  • Improvement: Set G711a as the preferred codec when no preferred codec is selected
  • Improvement: Refactored resource generation code to avoid code duplication
  • Deprecated: In certain cases, using this resource can negatively impact the codec negotiation of SIP trunks or SIP devices (M9787)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+
    • Communication Server module 2.0.0+ (If T.38 support is required)

Version 1.6.0 - Deprecated
  • Feature: Added T.38 support.
  • Deprecated: In certain cases, using this resource can negatively impact the codec negotiation of SIP trunks or SIP devices (M9787)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+
    • Communication Server module 2.0.0+ (If T.38 support is required)

Version 1.5.0 - Deprecated
  • Feature: Added active-active support (M0003637)
  • Deprecated: In certain cases, using this resource can negatively impact the codec negotiation of SIP trunks or SIP devices (M9787)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+

Version 1.4.0 - Deprecated
  • Bugfix: SIP and RTP ToS/DiffServ values not correct (M0004641)
  • Deprecated: In certain cases, using this resource can negatively impact the codec negotiation of SIP trunks or SIP devices (M9787)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+

Version 1.3.0 - Deprecated
  • Feature: Added SIP overload control settings
  • Deprecated: In certain cases, using this resource can negatively impact the codec negotiation of SIP trunks or SIP devices (M9787)
  • Dependency:
    • Asterisk-1.2.x module v2.30+
    • Cisco ATA support v1.4.0+

Version 1.2.0 - Deprecated
  • Bugfix: RTP Port for modules 2, 3 and 4 fixed
  • Improvement: Default Sip Ports changed to 5060,5061,5062,5063
  • Deprecated: In certain cases, using this resource can negatively impact the codec negotiation of SIP trunks or SIP devices (M9787)

Version 1.1 - Deprecated
  • Beta version of SPA8000 resource

Version 1.0 - Deprecated
  • Beta version of SPA8000 resource

Resource configuration interface

GUI unavailable.

Resource configuration parameters

  • DialTone: Dial tone configuration
  • BusyTone: Busy tone configuration
  • Reorder Tone: Reorder tone configuration
  • Off Hook Warning Tone: Reorder Tone configuration
  • Ring Back Tone: Ring Back Tone configuration
  • Line configuration: The ATA can buffer incoming voice packets to minimize the impact of variable network delays. This process is known as jitter buffering. The size of the jitter buffer adjusts to changing network conditions. The ATA has a Network Jitter Level control setting for each line of service. The jitter level determines how aggressively the ATA tries to shrink the jitter buffer over time to achieve a lower overall delay. If the jitter level is higher, it shrinks more gradually. If jitter level is lower, it shrinks more quickly
    • Network_Jitter Level: Determines how jitter buffer size is adjusted by the ATA. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high. Default setting: high
    • Jitter Buffer Adjustment: Choose yes to enable or no to disable this feature. Default setting: yes
    • Call Waiting Serv: Enable Call Waiting Service. Default setting: No
    • Three Way Conf Serv: Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer. Default setting: No
    • Echo Canc Enable: Enable the use of the echo canceller, select yes. Otherwise, select no. Default setting: No. Disable this if a fax is connected on the line.
    • Echo Canc Adapt Enable: To enable the echo canceller to adapt, select yes. Otherwise, select no. Default setting: No. Disable this if a fax is connected on the line.
    • Echo Supp Enable: To enable the use of the echo suppressor, select yes. Otherwise, select no. Default setting: Yes. Disable this if a fax is connected on the line.
    • Preferred Codec: Preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following: G711u, G711a, G726-32, or G729a. Default setting: G711u
    • Use Pref Codec Only: To use only the preferred codec for all calls, select yes. (The call fails if the far end does not support this codec.) Otherwise, select no. Default setting: no
    • Hotline Number: Number dialed in case of hotline. Indicate nothing to don't configure hotline.
    • Hotline dialing delay: Used only in case of hotline configured.
    • SIP ToS/DiffServ Value: TOS/DiffServ field value in UDP IP packets carrying a SIP message. Default setting: 0x68.
    • RTP ToS/DiffServ Value: TOS/DiffServ field value in UDP IP packets carrying RTP data. Default setting: 0xb8
    • SIP CoS Value: CoS value for SIP messages. Default setting: 3.
    • RTP CoS Value: CoS value for RTP data. Default setting: 6.
    • SIP Port Value: Port number of the SIP message listening and transmission port. Default setting: 5060.
    • SIP Server: Indicate here the IP address where to register. By default, it will be the IP address of SOP 1.
  • Line: Indicate the line to configure.
  • T38: Indicate if T38 must be enbaled
  • RTP Parameters: Indicate the range of RTP UDP port
  • Interdigit Long Timer:Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed. Range: 064 seconds. This setting can only be used in the first line identity and as global accross the device. Default setting: 10.
  • Interdigit Short Timer:Short timeout between entering digits when dialing. The Interdigit_Short_Timer specifies the default maximum time (in seconds) allowed between dialed digits, when at least one candidate digit sequence is complete as dialed. Range: 064 seconds. This setting can only be used in the first line identity and as global accross the device. Default setting: 3.
  • Dialplan: The Dialplan parameter contains the actual dial plan scripts for each of the identities. The default setting is good for an installation in Belgium with a 0 prefix to dial outside and 4 digit internal extensions. For more information concerning the dialplan parameter, please refer to the excellent documentation of Cisco. Default value: ([1-9]xxxS0|1[01]xS0|1[2]xx|004[6789]xxxxxxxS0|00[789]xxxxxxxS0|00[1-35-9]xxxxxxxS0|004xxxxxxxS2|000xx.)
  • DNS settings:
    • Whether to use DNS SRV lookup for Proxy and Outbound Proxy. Default setting: no.
    • If enabled, the ATA will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name. Default setting: no.

  • SIP server feature:
    • SIP Overload Control: 'Default' has currently the same function as 'Disabled' (planned for future use), 'Enable' to use the settings defined in this resource and 'Disabled' to deactivate the sip overload control
    • Max Call per period: The maximum number of call allowed during polling period. If the maximum is reached the SIP requests will be rejected with the a SIP error 480 Temporarily unavailable'
    • Polling period: Period during which the maximum number of calls is to be checked. At the end of each period the overload control counters are reset.
    • SIP blacklisting: 'Default' to keep settings defined in asterisk module, 'Enable' to use the settings defined in this resource and 'Disabled' to deactivate the sip blacklisting
    • Blacklisting time after overload: The time after the overload during which the IP address of incoming SIP requests will be blacklisted
    • IP blacklisted time: The time during which the IP address will remain blacklisted
    • Register Expires:
    • SIP reinvite: Yes to enable reinvite, Yo to disable reinvite. By default it will use the communication server settings.
    • SIP transport: Default is UDP. Select TCP to use TCP, UDP to use UDP and TLS to Use encrypted signaling between phone and SOP.

  • Internet access options
    • Time Configuration:
      • NTP Server 1: The primary NTP server, default is the SOP's IP address
      • NTP Server 2: The secondary NTP server.
    • Other configuration:
      • Nat Keep Alive Interval: Define the interval between NOTIFY messages sent to the SOP when behind a NAT. If left empty the default value of 60 seconds will be used. (in seconds)

Example of doorphone integration:

In certain country, doorphone needs to have specific tone set to the country specific tones. For example Fastel doorphone requires Belgian tones:

  • Dialtone: 480@-19,620@-19;10(.25/.25/1+2)
  • Reorder Tone: 425@-5;10(.5/.5/1)
  • Off Hook Warning Tone: 425@-5;10(.5/.5/1)
  • Ring Back Tone: 425@-5;10(3/1/1)
  • Busy Tone: 425@-5;10(.5/.5/1)
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