<div>Fernando, boa noite.</div>
<div>Acredito até que possa ser problemas em sua configuração, uma dica.</div>
<div>Delete esse seu zapata.conf e configure um limpo, com apenas aquilo que realmente você for utilizar, acho que será melhor para você administrar.</div>
<div>Poste na lista novamente, sem o debug, acho que dá para vermos alguma coisa.</div>
<div> </div>
<div>Abraço e boa sorte</div>
<div> </div>
<div>Josué<br><br> </div>
<div><span class="gmail_quote">2006/6/9, Fernando Lujan <<a href="mailto:fernando.lujan@mandic.com.br">fernando.lujan@mandic.com.br</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Saudações,<br><br>Estava tentando configurar o Asterisk com o MFC/R2 e não tive sucesso.<br>Pedi para a operadora trocar a sinalização para ISDN.
<br><br>Removi tudo referente a asterisk e reinstalei novamente.<br><br>O servidor que estou me ligando através de um T1 crossover está<br>configurado como slave.<br><br>Quando tento fazer uma ligação para o servidor no qual estou ligando o
<br>asterisk recebo esta informação. (ha*CLI><br><br>Creio que seja um problema de sinalização/protocolo.<br><br>Alguma idéia do que possa ser? Obrigado.<br><br><br>-- Accepting UNAUTHENTICATED call from <a href="http://192.168.1.59">
192.168.1.59</a>:<br> > requested format = gsm,<br> > requested prefs = (),<br> > actual format = gsm,<br> > host prefs = (),<br> > priority = mine<br> -- Executing Answer("IAX2/123456-2", "") in new stack
<br> -- Executing Dial("IAX2/123456-2", "Zap/g1/1144882120") in new stack<br>-- Making new call for cr 32770<br> -- Requested transfer capability: 0x00 - SPEECH<br>> Protocol Discriminator: Q.931
(8) len=49<br>> Call Ref: len= 2 (reference 2/0x2) (Originator)<br>> Message type: SETUP (5)<br>> [04 03 80 90 a3]<br>> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer<br>capability: Speech (0)
<br>> Ext: 1 Trans mode/rate: 64kbps,<br>circuit-mode (16)<br>> Ext: 1 User information layer 1: A-Law (35)<br>> [18 03 a1 83 81]<br>> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
<br>Preferred Dchan: 0<br>> ChanSel: Reserved<br>> Ext: 1 Coding: 0 Number Specified Channel<br>Type: 3<br>> Ext: 1 Channel: 1 ]<br>> [28 08 4d 6f 7a 50 68 6f 6e 65]
<br>> Display (len= 8) [ MozPhone ]<br>> [6c 08 00 80 39 39 39 39 39 39]<br>> Calling Number (len=10) [ Ext: 0 TON: Unknown Number Type (0) NPI:<br>Unknown Number Plan (0)<br>> Presentation: Presentation permitted, user
<br>number not screened (0) '999999' ]<br>> [70 0b 80 31 31 34 34 38 38 32 31 32 30]<br>> Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI:<br>Unknown Number Plan (0) '1144882120' ]<br>> [a1]<br>
> Sending Complete (len= 1)<br> -- Called g1/1144882120<br> == Primary D-Channel on span 1 down<br>Jun 9 14:41:22 WARNING[6430]: chan_zap.c:2290 pri_find_dchan: No<br>D-channels available! Using Primary channel 16 as D-channel anyway!
<br>NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated,<br>peerstate Overlap sending<br>> Protocol Discriminator: Q.931 (8) len=9<br>> Call Ref: len= 2 (reference 2/0x2) (Originator)<br>> Message type: DISCONNECT (69)
<br>> [08 02 81 90]<br>> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0<br>Location: Private network serving the local user (1)<br>> Ext: 1 Cause: Unknown (16), class = Normal Event (1) ]
<br>NEW_HANGUP DEBUG: Destroying the call, ourstate Disconnect Request,<br>peerstate Disconnect Indication<br> -- Hungup 'Zap/1-1'<br> == Everyone is busy/congested at this time (1:0/0/1)<br> == Auto fallthrough, channel 'IAX2/123456-2' status is 'CHANUNAVAIL'
<br> == Primary D-Channel on span 1 down<br>Jun 9 14:41:25 WARNING[6430]: chan_zap.c:2290 pri_find_dchan: No<br>D-channels available! Using Primary channel 16 as D-channel anyway!<br> > cdr_odbc: Query Successful!
<br> -- Hungup 'IAX2/123456-2'<br>bilitei o debug do span 1)<br><br>*CLI> -- Accepting UNAUTHENTICATED call from <a href="http://192.168.1.59">192.168.1.59</a>:<br> > requested format = gsm,<br> > requested prefs = (),
<br> > actual format = gsm,<br> > host prefs = (),<br> > priority = mine<br> -- Executing Answer("IAX2/123456-2", "") in new stack<br> -- Executing Dial("IAX2/123456-2", "Zap/g1/1144882120") in new stack
<br>-- Making new call for cr 32770<br> -- Requested transfer capability: 0x00 - SPEECH<br>> Protocol Discriminator: Q.931 (8) len=49<br>> Call Ref: len= 2 (reference 2/0x2) (Originator)<br>> Message type: SETUP (5)
<br>> [04 03 80 90 a3]<br>> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer<br>capability: Speech (0)<br>> Ext: 1 Trans mode/rate: 64kbps,<br>circuit-mode (16)<br>> Ext: 1 User information layer 1: A-Law (35)
<br>> [18 03 a1 83 81]<br>> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,<br>Preferred Dchan: 0<br>> ChanSel: Reserved<br>> Ext: 1 Coding: 0 Number Specified Channel
<br>Type: 3<br>> Ext: 1 Channel: 1 ]<br>> [28 08 4d 6f 7a 50 68 6f 6e 65]<br>> Display (len= 8) [ MozPhone ]<br>> [6c 08 00 80 39 39 39 39 39 39]<br>> Calling Number (len=10) [ Ext: 0 TON: Unknown Number Type (0) NPI:
<br>Unknown Number Plan (0)<br>> Presentation: Presentation permitted, user<br>number not screened (0) '999999' ]<br>> [70 0b 80 31 31 34 34 38 38 32 31 32 30]<br>> Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI:
<br>Unknown Number Plan (0) '1144882120' ]<br>> [a1]<br>> Sending Complete (len= 1)<br> -- Called g1/1144882120<br> == Primary D-Channel on span 1 down<br>Jun 9 14:41:22 WARNING[6430]: chan_zap.c:2290 pri_find_dchan: No
<br>D-channels available! Using Primary channel 16 as D-channel anyway!<br>NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated,<br>peerstate Overlap sending<br>> Protocol Discriminator: Q.931 (8) len=9<br>
> Call Ref: len= 2 (reference 2/0x2) (Originator)<br>> Message type: DISCONNECT (69)<br>> [08 02 81 90]<br>> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0<br>Location: Private network serving the local user (1)
<br>> Ext: 1 Cause: Unknown (16), class = Normal Event (1) ]<br>NEW_HANGUP DEBUG: Destroying the call, ourstate Disconnect Request,<br>peerstate Disconnect Indication<br> -- Hungup 'Zap/1-1'<br> == Everyone is busy/congested at this time (1:0/0/1)
<br> == Auto fallthrough, channel 'IAX2/123456-2' status is 'CHANUNAVAIL'<br> == Primary D-Channel on span 1 down<br>Jun 9 14:41:25 WARNING[6430]: chan_zap.c:2290 pri_find_dchan: No<br>D-channels available! Using Primary channel 16 as D-channel anyway!
<br> > cdr_odbc: Query Successful!<br> -- Hungup 'IAX2/123456-2'<br><br><br><br><br>zaptel.conf<br><br>span=1,1,0,ccs,hdb3<br>bchan=1-15,17-31<br>dchan=16<br>loadzone=us<br>defaultzone=us<br><br><br>zapata.conf
<br><br>;<br>; Zapata telephony interface<br>;<br>; Configuration file<br>;<br>; You need to restart Asterisk to re-configure the Zap channel<br>; CLI> reload chan_zap.so<br>; will reload the configuration file,
<br>; but not all configuration options are<br>; re-configured during a reload.<br><br><br><br>[trunkgroups]<br>;<br>; Trunk groups are used for NFAS or GR-303 connections.<br>;<br>; Group: Defines a trunk group.
<br>; group => <trunkgroup>,<dchannel>[,<backup1>...]<br>;<br>; trunkgroup is the numerical trunk group to create<br>; dchannel is the zap channel which will have the<br>; d-channel for the trunk.
<br>; backup1 is an optional list of backup d-channels.<br>;<br>;trunkgroup => 1,24,48<br>;trunkgroup => 1,24<br>;<br>; Spanmap: Associates a span with a trunk group<br>; spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
<br>;<br>; zapspan is the zap span number to associate<br>; trunkgroup is the trunkgroup (specified above) for the mapping<br>; logicalspan is the logical span number within the trunk group<br>to use.
<br>; if unspecified, no logical span number is used.<br>;<br>;spanmap => 1,1,1<br>;spanmap => 2,1,2<br>;spanmap => 3,1,3<br>;spanmap => 4,1,4<br><br>[channels]<br>;<br>; Default language<br>
;<br>language=us<br>;<br>; Default context<br>;<br>context=incoming<br>;<br>; Switchtype: Only used for PRI.<br>;<br>; national: National ISDN 2 (default)<br>; dms100: Nortel DMS100<br>; 4ess: AT&T 4ESS
<br>; 5ess: Lucent 5ESS<br>; euroisdn: EuroISDN<br>; ni1: Old National ISDN 1<br>; qsig: Q.SIG<br>;<br>switchtype=euroisdn<br>;<br>; Some switches (AT&T especially) require network specific facility IE
<br>; supported values are currently 'none', 'sdn', 'megacom', 'accunet'<br>;<br>;nsf=none<br>;<br>; PRI Dialplan: Only RARELY used for PRI.<br>;<br>; unknown: Unknown<br>; private: Private ISDN<br>; local: Local ISDN
<br>; national: National ISDN<br>; international: International ISDN<br>;<br>pridialplan=unknown<br>;<br>; PRI Local Dialplan: Only RARELY used for PRI (sets the calling<br>number's numbering plan)<br>;<br>; unknown: Unknown
<br>; private: Private ISDN<br>; local: Local ISDN<br>; national: National ISDN<br>; international: International ISDN<br>;<br>prilocaldialplan=unknown<br>;<br>; PRI callerid prefixes based on the given TON/NPI (dialplan)
<br>; This is especially needed for euroisdn E1-PRIs<br>;<br>; sample 1 for Germany<br>;internationalprefix = 00<br>;nationalprefix = 0<br>;localprefix = 0711<br>;privateprefix = 07115678<br>;unknownprefix =<br>;<br>; sample 2 for Germany
<br>;internationalprefix = +<br>;nationalprefix = +49<br>;localprefix = +49711<br>;privateprefix = +497115678<br>;unknownprefix =<br>;<br>; PRI resetinterval: sets the time in seconds between restart of unused<br>; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like
<br>; channel restarts. so set the interval to a very long interval e.g.<br>100000000<br>; or 'never' to disable *entirely*.<br>;<br>;resetinterval = 3600<br>;<br>; Overlap dialing mode (sending overlap digits)<br>;<br>;overlapdial=yes
<br>;<br>; PRI Out of band indications.<br>; Enable this to report Busy and Congestion on a PRI using out-of-band<br>; notification. Inband indication, as used by Asterisk doesn't seem to work<br>; with all telcos.<br>;<br>
; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT<br>; inband: Signal Busy/Congestion using in-band tones<br>;<br>; priindication = outofband<br>;<br>; If you need to override the existing channels selection routine and
<br>force all<br>; PRI channels to be marked as exclusively selected, set this to yes.<br>; priexclusive = yes<br>;<br>; ISDN Timers<br>; All of the ISDN timers and counters that are used are configurable.<br>Specify<br>; the timer name, and its value (in ms for timers).
<br>;<br>; pritimer => t200,1000<br>; pritimer => t313,4000<br>;<br>; To enable transmission of facility-based ISDN supplementary services (such<br>; as caller name from CPE over facility), enable this option.<br>; facilityenable = yes
<br>;<br>;<br>; Signalling method (default is fxs). Valid values:<br>; em: E & M<br>; em_w: E & M Wink<br>; featd: Feature Group D (The fake, Adtran style, DTMF)<br>; featdmf: Feature Group D (The real thing, MF (domestic, US))
<br>; featdmf_ta: Feature Group D (The real thing, MF (domestic, US))<br>through<br>; a Tandem Access point<br>; featb: Feature Group B (MF (domestic, US))<br>; fxs_ls: FXS (Loop Start)
<br>; fxs_gs: FXS (Ground Start)<br>; fxs_ks: FXS (Kewl Start)<br>; fxo_ls: FXO (Loop Start)<br>; fxo_gs: FXO (Ground Start)<br>; fxo_ks: FXO (Kewl Start)<br>; pri_cpe: PRI signalling, CPE side
<br>; pri_net: PRI signalling, Network side<br>; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side<br>; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side<br>; sf: SF (Inband Tone) Signalling
<br>; sf_w: SF Wink<br>; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)<br>; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))<br>; sf_featb: SF Feature Group B (MF (domestic, US))
<br>; e911: E911 (MF) style signalling<br>;<br>; The following are used for Radio interfaces:<br>; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO<br>at the<br>; channel bank)
<br>; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO<br>at the<br>; channel bank)<br>; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS<br>at the<br>; channel bank)
<br>; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at<br>; the channel bank)<br>; em_rx: Receive audio/COR on an E&M interface (1-way)<br>; em_tx: Transmit audio/PTT on an E&M interface (1-way)
<br>; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M<br>interface<br>; (2-way)<br>; em_rxtx: Same as em_txrx (for our dyslexic friends)<br>; sf_rx: Receive audio/COR on an SF interface (1-way)
<br>; sf_tx: Transmit audio/PTT on an SF interface (1-way)<br>; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF<br>interface<br>; (2-way)<br>; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
<br>;<br>signalling=pri_net<br>;<br>; For Feature Group D Tandem access, to set the default CIC and OZZ use<br>these<br>; parameters:<br>;defaultozz=0000<br>;defaultcic=303<br>;<br>; A variety of timing parameters can be specified as well
<br>; Including:<br>; prewink: Pre-wink time (default 50ms)<br>; preflash: Pre-flash time (default 50ms)<br>; wink: Wink time (default 150ms)<br>; flash: Flash time (default 750ms)<br>; start: Start time (default 1500ms)
<br>; rxwink: Receiver wink time (default 300ms)<br>; rxflash: Receiver flashtime (default 1250ms)<br>; debounce: Debounce timing (default 600ms)<br>;<br>rxwink=300 ; Atlas seems to use long (250ms) winks
<br>;<br>; How long generated tones (DTMF and MF) will be played on the channel<br>; (in miliseconds)<br>;toneduration=100<br>;<br>; Whether or not to do distinctive ring detection on FXO lines<br>;<br>;usedistinctiveringdetection=yes
<br><br>;<br>; Whether or not to use caller ID<br>;<br>usecallerid=yes<br>;<br>; Type of caller ID signalling in use<br>; bell = bell202 as used in US<br>; v23 = v23 as used in the UK<br>; dtmf = DTMF as used in Denmark, Sweden and Netherlands
<br>;<br>;cidsignalling=bell<br>;<br>; What signals the start of caller ID<br>; ring = a ring signals the start<br>; polarity = polarity reversal signals the start<br>;<br>;cidstart=ring<br>;<br>; Whether or not to hide outgoing caller ID (Override with *67 or *82)
<br>;<br>hidecallerid=no<br>;<br>; Whether or not to enable call waiting on FXO lines<br>;<br>callwaiting=yes<br>;<br>; Whether or not restrict outgoing caller ID (will be sent as ANI only, not<br>; available for the user)
<br>; Mostly use with FXS ports<br>;<br>;restrictcid=no<br>;<br>; Whether or not use the caller ID presentation for the outgoing call<br>that the<br>; calling switch is sending.<br>;<br>usecallingpres=yes<br>;<br>; Some countries (UK) have ring tones with different ring tones (ring-ring),
<br>; which means the callerid needs to be set later on, and not just after<br>; the first ring, as per the default.<br>;<br>;sendcalleridafter=1<br>;<br>;<br>; Support Caller*ID on Call Waiting<br>;<br>callwaitingcallerid=yes
<br>;<br>; Support three-way calling<br>;<br>threewaycalling=yes<br>;<br>; Support flash-hook call transfer (requires three way calling)<br>; Also enables call parking (overrides the 'canpark' parameter)<br>;<br>transfer=yes
<br>;<br>; Allow call parking<br>; ('canpark=no' is overridden by 'transfer=yes')<br>;<br>canpark=yes<br>;<br>; Support call forward variable<br>;<br>cancallforward=yes<br>;<br>; Whether or not to support Call Return (*69)
<br>;<br>callreturn=yes<br>;<br>; Stutter dialtone support: If a mailbox is specified without a voicemail<br>; context, then when voicemail is received in a mailbox in the default<br>; voicemail context in voicemail.conf, taking the phone off hook will
<br>cause a<br>; stutter dialtone instead of a normal one.<br>;<br>; If a mailbox is specified *with* a voicemail context, the same will result<br>; if voicemail recieved in mailbox in the specified voicemail context.<br>
;<br>; for default voicemail context, the example below is fine:<br>;<br>;mailbox=1234<br>;<br>; for any other voicemail context, the following will produce the<br>stutter tone:<br>;<br>;mailbox=1234@context<br>;<br>; Enable echo cancellation
<br>; Use either "yes", "no", or a power of two from 32 to 256 if you wish to<br>; actually set the number of taps of cancellation.<br>;<br>echocancel=yes<br>;<br>; Generally, it is not necessary (and in fact undesirable) to echo
<br>cancel when<br>; the circuit path is entirely TDM. You may, however, reverse this behavior<br>; by enabling the echo cancel during pure TDM bridging below.<br>;<br>echocancelwhenbridged=yes<br>;<br>; In some cases, the echo canceller doesn't train quickly enough and there
<br>; is echo at the beginning of the call. Enabling echo training will cause<br>; asterisk to briefly mute the channel, send an impulse, and use the impulse<br>; response to pre-train the echo canceller so it can start out with a much
<br>; closer idea of the actual echo. Value may be "yes", "no", or a number of<br>; milliseconds to delay before training (default = 400)<br>;<br>echotraining=yes<br>echotraining=800<br>;<br>; If you are having trouble with DTMF detection, you can relax the DTMF
<br>; detection parameters. Relaxing them may make the DTMF detector more<br>likely<br>; to have "talkoff" where DTMF is detected when it shouldn't be.<br>;<br>;relaxdtmf=yes<br>;<br>; You may also set the default receive and transmit gains (in dB)
<br>;<br>rxgain=0.0<br>txgain=0.0<br>;<br>; Logical groups can be assigned to allow outgoing rollover. Groups range<br>; from 0 to 63, and multiple groups can be specified.<br>;<br>group=1<br>;<br>; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
<br>; and it is a member of a group which is one of your pickup groups, then<br>; you can answer it by picking up and dialing *8#. For simple offices, just<br>; make these both the same<br>;<br>callgroup=1<br>pickupgroup=1
<br><br>;<br>; Specify whether the channel should be answered immediately or if the<br>simple<br>; switch should provide dialtone, read digits, etc.<br>;<br>immediate=no<br>;<br>; Specify whether flash-hook transfers to 'busy' channels should complete or
<br>; return to the caller performing the transfer (default is yes).<br>;<br>;transfertobusy=no<br>;<br>; CallerID can be set to "asreceived" or a specific number if you want to<br>; override it. Note that "asreceived" only applies to trunk interfaces.
<br>;<br>;callerid=2564286000<br>;<br>; AMA flags affects the recording of Call Detail Records. If specified<br>; it may be 'default', 'omit', 'billing', or 'documentation'.<br>;<br>;amaflags=default<br>;<br>; Channels may be associated with an account code to ease
<br>; billing<br>;<br>;accountcode=lss0101<br>;<br>; ADSI (Analog Display Services Interface) can be enabled on a per-channel<br>; basis if you have (or may have) ADSI compatible CPE equipment<br>;<br>;adsi=yes<br>;<br>; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
<br>; etc, it can be useful to perform busy detection either in an effort to<br>; detect hangup or for detecting busies. This enables listening for<br>; the beep-beep busy pattern.<br>;<br>;busydetect=yes<br>;<br>; If busydetect is enabled, it is also possible to specify how many busy
<br>tones<br>; to wait for before hanging up. The default is 4, but better results<br>can be<br>; achieved if set to 6 or even 8. Mind that the higher the number, the more<br>; time that will be needed to hangup a channel, but lowers the probability
<br>; that you will get random hangups.<br>;<br>;busycount=4<br>;<br>; If busydetect is enabled, it is also possible to specify the cadence<br>of your<br>; busy signal. In many countries, it is 500msec on, 500msec off. Without
<br>; busypattern specified, we'll accept any regular sound-silence pattern that<br>; repeats <busycount> times as a busy signal. If you specify busypattern,<br>; then we'll further check the length of the sound (tone) and silence, which
<br>; will further reduce the chance of a false positive.<br>;<br>;busypattern=500,500<br>;<br>; NOTE: In the Asterisk Makefile you'll find further options to tweak<br>the busy<br>; detector. If your country has a busy tone with the same length tone and
<br>; silence (as many countries do), consider defining the<br>; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.<br>;<br>; Use a polarity reversal to mark when a outgoing call is answered by the<br>; remote party.<br>;<br>;answeronpolarityswitch=yes
<br>;<br>; In some countries, a polarity reversal is used to signal the<br>disconnect of a<br>; phone line. If the hanguponpolarityswitch option is selected, the<br>call will<br>; be considered "hung up" on a polarity reversal.
<br>;<br>;hanguponpolarityswitch=yes<br>;<br>; On trunk interfaces (FXS) it can be useful to attempt to follow the<br>progress<br>; of a call through RINGING, BUSY, and ANSWERING. If turned on, call<br>; progress attempts to determine answer, busy, and ringing on phone lines.
<br>; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,<br>; so don't count on it being very accurate.<br>;<br>; Few zones are supported at the time of this writing, but may be selected<br>; with "progzone"
<br>;<br>; This feature can also easily detect false hangups. The symptoms of this is<br>; being disconnected in the middle of a call for no reason.<br>;<br>;callprogress=yes<br>;progzone=us<br>;<br>; FXO (FXS signalled) devices must have a timeout to determine if there
<br>was a<br>; hangup before the line was answered. This value can be tweaked to shorten<br>; how long it takes before Zap considers a non-ringing line to have hungup.<br>;<br>;ringtimeout=8000<br>;<br>; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
<br>;<br>;pulsedial=yes<br>;<br>; For fax detection, uncomment one of the following lines. The default<br>is *OFF*<br>;<br>;faxdetect=both<br>;faxdetect=incoming<br>;faxdetect=outgoing<br>;faxdetect=no<br>;<br>; Select which class of music to use for music on hold. If not specified
<br>; then the default will be used.<br>;<br>musiconhold=default<br>;<br>; PRI channels can have an idle extension and a minunused number. So<br>long as<br>; at least "minunused" channels are idle, chan_zap will try to call
<br>"idledial"<br>; on them, and then dump them into the PBX in the "idleext" extension (which<br>; is of the form exten@context). When channels are needed the "idle" calls<br>; are disconnected (so long as there are at least "minidle" calls still
<br>; running, of course) to make more channels available. The primary use of<br>; this is to create a dynamic service, where idle channels are bundled<br>through<br>; multilink PPP, thus more efficiently utilizing combined voice/data
<br>services<br>; than conventional fixed mappings/muxings.<br>;<br>;idledial=6999<br>;idleext=6999@dialout<br>;minunused=2<br>;minidle=1<br>;<br>; Configure jitter buffers in zapata (each one is 20ms, default is 4)<br>;<br>
;jitterbuffers=4<br>;<br>; You can define your own custom ring cadences here. You can define up to 8<br>; pairs. If the silence is negative, it indicates where the callerid<br>spill is<br>; to be placed. Also, if you define any custom cadences, the default
<br>cadences<br>; will be turned off.<br>;<br>; Syntax is: cadence=ring,silence[,ring,silence[...]]<br>;<br>; These are the default cadences:<br>;<br>;cadence=125,125,2000,-4000<br>;cadence=250,250,500,1000,250,250,500,-4000
<br>;cadence=125,125,125,125,125,-4000<br>;cadence=1000,500,2500,-5000<br>;<br>; Each channel consists of the channel number or range. It inherits the<br>; parameters that were specified above its declaration.<br>;<br>; For GR-303, CRV's are created like channels except they must start
<br>with the<br>; trunk group followed by a colon, e.g.:<br>;<br>; crv => 1:1<br>; crv => 2:1-2,5-8<br>;<br>;<br>;callerid="Green Phone"<(256) 428-6121><br>;channel => 1<br>;callerid="Black Phone"<(256) 428-6122>
<br>;channel => 2<br>;callerid="CallerID Phone" <(256) 428-6123><br>;callerid="CallerID Phone" <(630) 372-1564><br>;callerid="CallerID Phone" <(256) 704-4666><br>;channel => 3
<br>;callerid="Pac Tel Phone" <(256) 428-6124><br>;channel => 4<br>;callerid="Uniden Dead" <(256) 428-6125><br>;channel => 5<br>;callerid="Cortelco 2500" <(256) 428-6126>
<br>;channel => 6<br>;callerid="Main TA 750" <(256) 428-6127><br>;channel => 44<br>;<br>; For example, maybe we have some other channels which start out in a<br>; different context and use E & M signalling instead.
<br>;<br>;context=remote<br>;sigalling=em<br>;channel => 15<br>;channel => 16<br><br>;signalling=em_w<br>;<br>; All those in group 0 I'll use for outgoing calls<br>;<br>; Strip most significant digit (9) before sending
<br>;<br>;stripmsd=1<br>;callerid=asreceived<br>;group=0<br>;signalling=fxs_ls<br>;channel => 45<br><br>;signalling=fxo_ls<br>;group=1<br>;callerid="Joe Schmoe" <(256) 428-6131><br>;channel => 25<br>
;callerid="Megan May" <(256) 428-6132><br>;channel => 26<br>;callerid="Suzy Queue" <(256) 428-6233><br>;channel => 27<br>;callerid="Larry Moe" <(256) 428-6234><br>;channel => 28
<br>;<br>; Sample PRI (CPE) config: Specify the switchtype, the signalling as either<br>; pri_cpe or pri_net for CPE or Network termination, and generally you will<br>; want to create a single "group" for all channels of the PRI.
<br>;<br>; switchtype = national<br>; signalling = pri_cpe<br>; group = 2<br>; channel => 1-23<br><br>;<br><br>; Used for distintive ring support for x100p.<br>; You can see the dringX patterns is to set any one of the
<br>dringXcontext fields<br>; and they will be printed on the console when an inbound call comes in.<br>;<br>;dring1=95,0,0<br>;dring1context=internal1<br>;dring2=325,95,0<br>;dring2context=internal2<br>; If no pattern is matched here is where we go.
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