PRI & Traditional T1 signalling methods,switchtype

Discussion in 'General' started by 6string, May 1, 2009.

  1. 6string

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    I have searched high and low for this information, it is very useful for configuring traditional T1's and different variations of PRI's particularly different signaling methods. Very useful for connecting Elastix to legacy PBX's that support T1 but not PRI.
    I'm sure some of this is not applicable for Elastix but it has helped me
    I hope this helps you too!!


    Switchtype: Only used for PRI.

    national: National ISDN 2 (default)
    dms100: Nortel DMS100
    4ess: AT&T 4ESS
    5ess: Lucent 5ESS
    euroisdn: EuroISDN
    ni1: Old National ISDN 1
    qsig: Q.SIG

    switchtype=national


    Overlap dialing mode (sending overlap digits)

    overlapdial=yes

    PRI Out of band indications.

    Enable this to report Busy and Congestion on a PRI using out-of-band notification. Inband indication, as used by Asterisk doesn't seem to work with all telcos.

    outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
    inband: Signal Busy/Congestion using in-band tones

    priindication = outofband

    If you need to override the existing channels selection routine and force all PRI channels to be marked as exclusively selected, set this to yes.
    priexclusive = yes

    ISDN Timers
    All of the ISDN timers and counters that are used are configurable. Specify the timer name, and its value (in ms for timers).

    pritimer => t200,1000
    pritimer => t313,4000

    To enable transmission of facility-based ISDN supplementary services (such as caller name from CPE over facility), enable this option.
    facilityenable = yes


    Signalling method (default is fxs). Valid values:
    em: E & M
    em_w: E & M Wink
    featd: Feature Group D (The fake, Adtran style, DTMF)
    featdmf: Feature Group D (The real thing, MF (domestic, US))
    featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
    a Tandem Access point
    featb: Feature Group B (MF (domestic, US))
    fxs_ls: FXS (Loop Start)
    fxs_gs: FXS (Ground Start)
    fxs_ks: FXS (Kewl Start)
    fxo_ls: FXO (Loop Start)
    fxo_gs: FXO (Ground Start)
    fxo_ks: FXO (Kewl Start)
    pri_cpe: PRI signalling, CPE side
    pri_net: PRI signalling, Network side
    gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
    gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side
    sf: SF (Inband Tone) Signalling
    sf_w: SF Wink
    sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
    sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
    sf_featb: SF Feature Group B (MF (domestic, US))
    e911: E911 (MF) style signalling

    A variety of timing parameters can be specified as well
    Including:
    prewink: Pre-wink time (default 50ms)
    preflash: Pre-flash time (default 50ms)
    wink: Wink time (default 150ms)
    flash: Flash time (default 750ms)
    start: Start time (default 1500ms)
    rxwink: Receiver wink time (default 300ms)
    rxflash: Receiver flashtime (default 1250ms)
    debounce: Debounce timing (default 600ms)
    rxwink=300 ; Atlas seems to use long (250ms) winks

    How long generated tones (DTMF and MF) will be played on the channel (in miliseconds)
    toneduration=100


    Whether or not to do distinctive ring detection on FXO lines

    usedistinctiveringdetection=yes

    Whether or not to use caller ID

    usecallerid=yes

    Type of caller ID signalling in use
    bell = bell202 as used in US
    v23 = v23 as used in the UK
    dtmf = DTMF as used in Denmark, Sweden and Netherlands

    cidsignalling=bell

    What signals the start of caller ID
    ring = a ring signals the start
    polarity = polarity reversal signals the start

    cidstart=ring

    Whether or not to hide outgoing caller ID (Override with *67 or *82)

    hidecallerid=no

    Whether or not to enable call waiting on FXO lines

    callwaiting=yes

    Whether or not restrict outgoing caller ID (will be sent as ANI only, not
    available for the user)
    Mostly use with FXS ports

    restrictcid=no

    Whether or not use the caller ID presentation for the outgoing call that the
    calling switch is sending.

    usecallingpres=yes

    Some countries (UK) have ring tones with different ring tones (ring-ring), which means the callerid needs to be set later on, and not just after
    the first ring, as per the default.

    sendcalleridafter=1


    Support Caller*ID on Call Waiting

    callwaitingcallerid=yes

    Support three-way calling

    threewaycalling=yes

    Support flash-hook call transfer (requires three way calling). Also enables call parking (overrides the 'canpark' parameter)

    transfer=yes



    mailbox=1234
    for any other voicemail context, the following will produce the stutter tone:

    mailbox=1234@context

    Enable echo cancellation
    Use either "yes", "no", or a power of two from 32 to 256 if you wish to actually set the number of taps of cancellation.

    echocancel=yes

    Generally, it is not necessary (and in fact undesirable) to echo cancel when the circuit path is entirely TDM. You may, however, reverse this behavior by enabling the echo cancel during pure TDM bridging below.

    echocancelwhenbridged=yes

    In some cases, the echo canceller doesn't train quickly enough and there is echo at the beginning of the call. Enabling echo training will cause asterisk to briefly mute the channel, send an impulse, and use the impulse response to pre-train the echo canceller so it can start out with a much closer idea of the actual echo. Value may be "yes", "no", or a number of milliseconds to delay before training (default = 400)

    echotraining=yes
    echotraining=800

    If you are having trouble with DTMF detection, you can relax the DTMF detection parameters. Relaxing them may make the DTMF detector more likely to have "talkoff" where DTMF is detected when it shouldn't be.

    relaxdtmf=yes

    You may also set the default receive and transmit gains (in dB)

    rxgain=0.0
    txgain=0.0

    Logical groups can be assigned to allow outgoing rollover. Groups range from 0 to 63, and multiple groups can be specified.
    group=1

    Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing and it is a member of a group which is one of your pickup groups, then you can answer it by picking up and dialing *8#. For simple offices, just make these both the same

    callgroup=1
    pickupgroup=1

    Specify whether the channel should be answered immediately or if the simple switch should provide dialtone, read digits, etc.

    immediate=no

    Specify whether flash-hook transfers to 'busy' channels should complete or return to the caller performing the transfer (default is yes).

    transfertobusy=no

    CallerID can be set to "asreceived" or a specific number if you want to override it. Note that "asreceived" only applies to trunk interfaces.

    callerid=2564286000

    ADSI (Analog Display Services Interface) can be enabled on a per-channel basis if you have (or may have) ADSI compatible CPE equipment

    adsi=yes

    On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D etc, it can be useful to perform busy detection either in an effort to detect hangup or for detecting busies. This enables listening for the beep-beep busy pattern.

    busydetect=yes

    If busydetect is enabled, it is also possible to specify how many busy tones to wait for before hanging up. The default is 4, but better results can be achieved if set to 6 or even 8. Mind that the higher the number, the more time that will be needed to hangup a channel, but lowers the probability that you will get random hangups.

    busycount=4

    If busydetect is enabled, it is also possible to specify the cadence of your busy signal. In many countries, it is 500msec on, 500msec off. Without busypattern specified, we'll accept any regular sound-silence pattern that repeats <busycount> times as a busy signal. If you specify busypattern, then we'll further check the length of the sound (tone) and silence, which will further reduce the chance of a false positive.

    busypattern=500,500

    NOTE: In the Asterisk Makefile you'll find further options to tweak the busy detector. If your country has a busy tone with the same length tone and silence (as many countries do), consider defining the
    -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.

    Use a polarity reversal to mark when a outgoing call is answered by the remote party.
    answeronpolarityswitch=yes

    In some countries, a polarity reversal is used to signal the disconnect of a phone line. If the hanguponpolarityswitch option is selected, the call will be considered "hung up" on a polarity reversal.

    hanguponpolarityswitch=yes

    On trunk interfaces (FXS) it can be useful to attempt to follow the progress of a call through RINGING, BUSY, and ANSWERING. If turned on, call progress attempts to determine answer, busy, and ringing on phone lines.

    This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, so don't count on it being very accurate.

    Few zones are supported at the time of this writing, but may be selected with "progzone"

    This feature can also easily detect false hangups. The symptoms of this is being disconnected in the middle of a call for no reason.

    callprogress=yes
    progzone=us

    FXO (FXS signalled) devices must have a timeout to determine when there was a hangup before the line was answered. This value can be tweaked to shorten how long it takes before Zap considers a non-ringing line to have hungup.

    ringtimeout=8000

    For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
    pulsedial=yes

    For fax detection, uncomment one of the following lines. The default is *OFF*

    faxdetect=both
    faxdetect=incoming
    faxdetect=outgoing
    faxdetect=no

    Select which class of music to use for music on hold. If not specified then the default will be used.

    musiconhold=default

    PRI channels can have an idle extension and a minunused number. So long as at least "minunused" channels are idle, chan_zap will try to call "idledial" on them, and then dump them into the PBX in the "idleext" extension (which is of the form exten@context). When channels are needed the "idle" calls are disconnected (so long as there are at least "minidle" calls still running, of course) to make more channels available. The primary use of this is to create a dynamic service, where idle channels are bundled through multilink PPP, thus more efficiently utilizing combined voice/data services than conventional fixed mappings/muxings.

    idledial=6999
    idleext=6999@dialout
    minunused=2
    minidle=1

    Configure jitter buffers in zapata (each one is 20ms, default is 4)

    jitterbuffers=4


    For example, maybe we have some other channels which start out in a different context and use E & M signalling instead.

    context=remote
    sigalling=em
    channel => 15
    channel => 16

    signalling=em_w

    All those in group 0 I'll use for outgoing calls

    Strip most significant digit (9) before sending

    stripmsd=1
    callerid=asreceived
    group=0
    signalling=fxs_ls
    channel => 45
    signalling=fxo_ls
    group=1
    callerid="Joe Schmoe" <(256) 428-6131>
    channel => 25
    callerid="Megan May" <(256) 428-6132>
    channel => 26
    callerid="Suzy Queue" <(256) 428-6233>
    channel => 27
    callerid="Larry Moe" <(256) 428-6234>
    channel => 28

    Sample PRI (CPE) config: Specify the switchtype, the signalling as either pri_cpe or pri_net for CPE or Network termination, and generally you will want to create a single "group" for all channels of the PRI.

    switchtype = national
    signalling = pri_cpe
    group = 2
    channel => 1-23
     
  2. johnme

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