[Asterisk-Users] (no subject)
chad at mai-telecom.com
Wed Jun 16 17:16:31 MST 2004
We are using the Digium 405PP card, and getting the following messages:
Jun 16 16:16:17 NOTICE: chan_zap.c:6832 pri_dchannel: PRI got event:
6 on Primary D-channel of span 1
Jun 16 16:16:17 NOTICE: chan_zap.c:6832 pri_dchannel: PRI got event:
8 on Primary D-channel of span 1
My config file is below. We are trying to set up D-Channel on channel 24,
1-23 in trunk group 1, 5ESS for switch type, and Network for ISDN. We are
connecting to a Cisco AS5300.
Please let me know if you can help me isolate the problem.
; Zapata telephony interface
; Configuration file
; Default language
; Default context
; 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
; PRI Dialplan: Only RARELY used for PRI.
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
; Overlap dialing mode (sending overlap digits)
; 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))
; 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
; 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))
; The following are used for Radio interfaces:
; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at the
; em_rx: Receive audio/COR on an E&M interface (1-way)
; em_tx: Transmit audio/PTT on an E&M interface (1-way)
; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
; em_rxtx: same as em_txrx (for our dyslexic friends)
; sf_rx: Receive audio/COR on an SF interface (1-way)
; sf_tx: Transmit audio/PTT on an SF interface (1-way)
; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
; sf_rxtx: same as sf_txrx (for our dyslexic friends)
; Whether or not to do distinctive ring detection on FXO lines
; Whether or not to use caller ID
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
; Whether or not to enable call waiting on FXO lines
; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
available for the user)
; Mostly use with FXS ports
; Whether or not use the caller ID presentation for the outgoing call that
the calling switch is sending
; Support Caller*ID on Call Waiting
; Support three-way calling
; Support flash-hook call transfer (requires three way calling)
; Support call forward variable
; Whether or not to support Call Return (*69)
; Stutter dialtone support: If a mailbox is specified, then when voicemail
; is received in that mailbox, taking the phone off hook will cause
; a stutter dialtone instead of a normal one
; 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.
; 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.
; 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.
; 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.
; You may also set the default receive and transmit gains (in dB)
; Logical groups can be assigned to allow outgoing rollover. Groups
; range from 0 to 31, and multiple groups can be specified.
; 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
; Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
; CallerID can be set to "asreceived" or a specific number
; if you want to override it. Note that "asreceived" only
; applies to trunk interfaces.
; AMA flags affects the recording of Call Detail Records. If specified
; it may be 'default', 'omit', 'billing', or 'documentation'.
; Channels may be associated with an account code to ease
; ADSI (Analog Display Services Interface) can be enabled on a per-channel
; basis if you have (or may have) ADSI compatible CPE equipment
; 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
; If busydetect is enabled, is also possible to specify how many
; busy tones to wait before hanging up. The default is 4, but
; better results can be achieved if set to 6 or even 8. Mind that
; higher the number, more time is needed to hangup a channel, but
; lower is probability to get random hangups
; On trunk interfaces (FXS) it can be useful to attempt to follow the
; 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. Also, it is ONLY configured for
; standard U.S. tones. This feature can also easily detect false hangups.
; The symptoms of this is being disconnected in the middle of a call for no
; Select which class of music to use for music on hold. If not specified
; then the default will be used.
; 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 at 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.
;idleext=6999 at dialout
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
; Each channel consists of the channel number or range. It
; inherits the parameters that were specified above its declaration
;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
; For example, maybe we have some other channels
; which start out in a different context and use
; E & M signalling instead.
;channel => 15
;channel => 16
; All those in group 0 I'll use for outgoing calls
; Strip most significant digit (9) before sending
;channel => 45
;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 = 5ess
signalling = pri_net
group = 1
channel => 1-23
; Used for distintive ring support for x100p.
; You can see the dringX patterns is to set any one of the dringXcontext
; and they will be printed on the console when an inbound call comes in.
; If no pattern is matched here is where we go.
;channel => 1
More information about the asterisk-users