Cisco SPA112 (SDSD)

Description

This resource allows to connect a Cisco SPA112 ATA.

Release notes

Resource configuration interface

Tone configuration format:

A tone is a mini-script of up to 127 characters that specifies the frequency, level and cadence parameters of a call progress tone. May contain up to 127 characters.

• Syntax: FreqScript;Z1[;Z2].
  • Z1 = D1(oni,1/offi,1/fi,1[,oni,2/offi,2/fi,2 [,oni,3/offi,3/fi,3[,oni,4/offi,4/fi,4[,oni,5/offi,5/fi,5[,oni,6/offi,6/fi,6]]]]]), where fi,j = n1[+n2]+n3[+n4[+n5[+n6]]]]] and 1 < nk < 6 indicates which of the frequency components given in the FreqScript are used in that segment; if more than one frequency component is used in a segment, the components are summed together.
  • FreqScript has the Syntax: F1@L1[,F2@L2[,F3@L3[,F4@L4[,F5@L5[,F6@L6]]]]. Where F1–F6 are frequency in Hz (unsigned integers only) and L1–L6 are corresponding levels in dBm (with up to 1 decimal places).

Cisco SPA112 : 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: 0–64 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: 0–64 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.)
• DHCP Server: The status of the DHCP server: Enabled or Disabled.
• DTMF mode: default (as indicated in communication server), RFC2833 or inband
• 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.
  • Use SRTP: Use encrypted voice packets between phone and SOP.
• SecondDialTone: Second dial tone configuration
• OutsideDialTone: Outside dial tone configuration
• PromptTone: Prompt tone configuration

• 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@-19;10(.5/.5/1)
• Off Hook Warning Tone: 425@-19;10(.5/.5/1)
• Ring Back Tone: 425@-19;10(3/1/1)
• Busy Tone: 425@-19;10(.5/.5/1)

Troubleshooting

The device does not register

Depending on the firmware version, the device may not reboot after fetching the first part of its configuration. In that case, the power light will be on without blinking for more than 5 minutes but the phones lights will be off. Simply unplug and replug the device power to reboot it. It should then download the right configuration and register to the SOP.

The audio stream is one way only (Versions prior to 1.13.2)

If you get one way audio, check the settings to ensure that you don't have the following set :

• Network_Jitter Level : low

This configuration will lead to one way audio in every case with this device. To solve the problem, set the "Network_Jitter Level" to another value.

Fax configuration with Passthru codec specific case

The use of G711 passthrough when sending or receiving fax with more than 5 page, you can get an issue where the sender close the fax communication. To fix this issue, we recommand configure the resource with following parameters set to specific values

• Line configuration
  • Network Jitter Level: very high
  • Preferred Codec: G711a
  • SIP ToS/DiffServ Value: 0x68
  • RTP ToS/DiffServ Value: 0x2E
  • SIP CoS Value: 3
  • RTP CoS Value: 5
• Misc. Settings
  • Fax passthru codec: G711a

Ensure that the preferred codec is the same as the Fax passthru codec.