All pastes #186 Raw Edit

Why dont I get CallerID

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#186 ·published 2004-06-07 06:10 UTC
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;; Zapata telephony interface;; Configuration file[channels];; Default language;;language=en;; Default context;context=incoming;; 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;switchtype=national;; PRI Dialplan:  Only RARELY used for PRI.;; unknown:        Unknown; private:        Private ISDN; local:          Local ISDN; national:	  National ISDN; international:  International ISDN;;pridialplan=national;; Overlap dialing mode (sending overlap digits);;overlapdial=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)); 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 channel bank); fxs_tx:  Transmit audio/PTT on an FXS loopstart interface (FXO at the channel bank); fxo_rx:  Receive audio/COR on an FXO loopstart interface (FXS at the channel bank); fxo_tx:  Transmit audio/PTT on an FXO groundstart interface (FXS at the channel bank); 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 (2-way); 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 (2-way); sf_rxtx: same as sf_txrx (for our dyslexic friends);signalling=fxo_ks;; A variety of timing parameters can be specified as well; Including:;    prewink:     Pre-wink time;    preflash:    Pre-flash time;    wink:        Wink time;    flash:       Flash time;    start:       Start time;    rxwink:      Receiver wink time;    rxflash:     Receiver flashtime;    debounce:    Debounce timing;rxwink=300		; Atlas seems to use long (250ms) winks;; Whether or not to do distinctive ring detection on FXO lines;;usedistinctiveringdetection=yes;; Whether or not to use caller ID;usecallerid=yes;; 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;; Support Caller*ID on Call Waiting;callwaitingcallerid=yes;; Support three-way calling;threewaycalling=yes;; Support flash-hook call transfer (requires three way calling);transfer=yes;; Support call forward variable;cancallforward=yes;; Whether or not to support Call Return (*69);callreturn=yes;; 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;;mailbox=1234;; 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.;;echotraining=yes;; 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=0.0txgain=0.0;; Logical groups can be assigned to allow outgoing rollover.  Groups; range from 0 to 31, 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=1pickupgroup=1;; Specify whether the channel should be answered immediately or; if the simple switch should provide dialtone, read digits, etc.;immediate=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=asreceived;; AMA flags affects the recording of Call Detail Records.  If specified; it may be 'default', 'omit', 'billing', or 'documentation'.;;amaflags=default;; Channels may be associated with an account code to ease; billing;;accountcode=lss0101;; 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;;busydetect=yes;; 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;;busycount=4;; 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.  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; reason.;;callprogress=yes;; 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;; 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.;;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;;  Used for distintive ring support for x100p.;  You can see the dringX patterns is to set any one of the dringXcontext fields;  and they will be printed on the console when an inbound call comes in.;;dring1=95,0,0 ;dring1context=internal1 ;dring2=325,95,0 ;dring2context=internal2 ; If no pattern is matched here is where we go.;context=default;channel => 1 context=incomingsignalling=fxs_kschannel=>2