From 9e1d0d917f0516fe76299e3af77a610a726d7ffa Mon Sep 17 00:00:00 2001 From: kpfleming Date: Fri, 11 Jul 2008 16:08:03 +0000 Subject: new installations should be using DAHDI instead of Zaptel, so the sample config file is now chan_dahdi.conf instead of zapata.conf also, convert remaining references to zapata.conf in various places git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@130042 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/chan_dahdi.conf.sample | 675 +++++++++++++++++++++++++++++++++++++++++ configs/extensions.ael.sample | 2 +- configs/extensions.conf.sample | 2 +- configs/sla.conf.sample | 2 +- configs/smdi.conf.sample | 2 +- configs/zapata.conf.sample | 675 ----------------------------------------- 6 files changed, 679 insertions(+), 679 deletions(-) create mode 100644 configs/chan_dahdi.conf.sample delete mode 100644 configs/zapata.conf.sample (limited to 'configs') diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample new file mode 100644 index 000000000..70bc8a208 --- /dev/null +++ b/configs/chan_dahdi.conf.sample @@ -0,0 +1,675 @@ +; +; DAHDI telephony interface +; +; Configuration file +; +; You need to restart Asterisk to re-configure the DAHDI channels +; CLI> reload chan_dahdi.so +; will reload the configuration file, +; but not all configuration options are +; re-configured during a reload. + + + +[trunkgroups] +; +; Trunk groups are used for NFAS or GR-303 connections. +; +; Group: Defines a trunk group. +; trunkgroup => ,[,...] +; +; trunkgroup is the numerical trunk group to create +; dchannel is the DAHDI channel which will have the +; d-channel for the trunk. +; backup1 is an optional list of backup d-channels. +; +;trunkgroup => 1,24,48 +;trunkgroup => 1,24 +; +; Spanmap: Associates a span with a trunk group +; spanmap => ,[,] +; +; dahdispan is the DAHDI span number to associate +; trunkgroup is the trunkgroup (specified above) for the mapping +; logicalspan is the logical span number within the trunk group to use. +; if unspecified, no logical span number is used. +; +;spanmap => 1,1,1 +;spanmap => 2,1,2 +;spanmap => 3,1,3 +;spanmap => 4,1,4 + +[channels] +; +; Default language +; +;language=en +; +; Default context +; +context=default +; +; Switchtype: Only used for PRI. +; +; national: National ISDN 2 (default) +; dms100: Nortel DMS100 +; 4ess: AT&T 4ESS +; 5ess: Lucent 5ESS +; euroisdn: EuroISDN (also known as ETSI NET/5; Cisco calls this "primary-net5") +; ni1: Old National ISDN 1 +; qsig: Q.SIG +; +switchtype=national +; +; Some switches (AT&T especially) require network specific facility IE +; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet' +; +;nsf=none +; +; PRI Dialplan: Only RARELY used for PRI. +; +; unknown: Unknown +; private: Private ISDN +; local: Local ISDN +; national: National ISDN +; international: International ISDN +; dynamic: Dynamically selects the appropriate dialplan +; +;pridialplan=national +; +; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan) +; +; unknown: Unknown +; private: Private ISDN +; local: Local ISDN +; national: National ISDN +; international: International ISDN +; dynamic: Dynamically selects the appropriate dialplan +; +;prilocaldialplan=national +; +; PRI callerid prefixes based on the given TON/NPI (dialplan) +; This is especially needed for euroisdn E1-PRIs +; +; sample 1 for Germany +;internationalprefix = 00 +;nationalprefix = 0 +;localprefix = 0711 +;privateprefix = 07115678 +;unknownprefix = +; +; sample 2 for Germany +;internationalprefix = + +;nationalprefix = +49 +;localprefix = +49711 +;privateprefix = +497115678 +;unknownprefix = +; +; PRI resetinterval: sets the time in seconds between restart of unused +; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like +; channel restarts. so set the interval to a very long interval e.g. 100000000 +; or 'never' to disable *entirely*. +; +;resetinterval = 3600 +; +; Overlap dialing mode (sending overlap digits) +; +;overlapdial=yes +; +; Allow inband audio (progress) when a call is RELEASEd by the far end of a PRI +; +;inbandrelease=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). +; K: Layer 2 max number of outstanding unacknowledged I frames (default 7) +; N200: Layer 2 max number of retransmissions of a frame (default 3) +; T200: Layer 2 max time before retransmission of a frame (default 1000 ms) +; T203: Layer 2 max time without frames being exchanged (default 10000 ms) +; T305: Wait for DISCONNECT acknowledge (default 30000 ms) +; T308: Wait for RELEASE acknowledge (default 4000 ms) +; T309: Maintain active calls on Layer 2 disconnection (default -1, Asterisk clears calls) +; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s +; May vary in other ISDN standards (Q.931 1993 : 90000 ms) +; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms) +; +; 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)) +; fgccama Feature Group C-CAMA (DP DNIS, MF ANI) +; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI) +; 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 +; +; 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_ls +; +; If you have an outbound signalling format that is different from format +; specified above (but compatible), you can specify outbound signalling format, +; (see below). The 'signalling' format specified will be the inbound signalling +; format. If you only specify 'signalling', then it will be the format for +; both inbound and outbound. +; +; signalling=featdmf +; outsignalling=featb +; +; For Feature Group D Tandem access, to set the default CIC and OZZ use these +; parameters: +;defaultozz=0000 +;defaultcic=303 +; +; 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 milliseconds) +;toneduration=100 +; +; Whether or not to do distinctive ring detection on FXO lines +; +;usedistinctiveringdetection=yes +;distinctiveringaftercid=yes ; enable dring detection after callerid for those countries like Australia + ; where the ring cadence is changed *after* the callerid spill. +; +; 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 +; v23_jp = v23 as used in Japan +; dtmf = DTMF as used in Denmark, Sweden and Netherlands +; smdi = Use SMDI for callerid. Requires SMDI to be enabled (usesmdi). +; +;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 internal extensions +; With this set to 'yes', busy extensions will hear the call-waiting +; tone, and can use hook-flash to switch between callers. The Dial() +; app will not return the "BUSY" result for extensions. +; +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. +; See doc/callingpres.txt +; +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 +; +; For FXS ports (either direct analog or over T1/E1): +; Support flash-hook call transfer (requires three way calling) +; Also enables call parking (overrides the 'canpark' parameter) +; +; For digital ports using ISDN PRI protocols: +; Support switch-side transfer (called 2BCT, RLT or other names) +; This setting must be enabled on both ports involved, and the +; 'facilityenable' setting must also be enabled to allow sending +; the transfer to the ISDN switch, since it sent in a FACILITY +; message. +; +transfer=yes +; +; Allow call parking +; ('canpark=no' is overridden by 'transfer=yes') +; +canpark=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 without a voicemail +; context, then when voicemail is received in a mailbox in the default +; voicemail context in voicemail.conf, taking the phone off hook will cause a +; stutter dialtone instead of a normal one. +; +; If a mailbox is specified *with* a voicemail context, the same will result +; if voicemail received in mailbox in the specified voicemail context. +; +; for default voicemail context, the example below is fine: +; +;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. +; +; Note that when setting the number of taps, the number 256 does not translate +; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms. +; +; Note that if any of your DAHDI cards have hardware echo cancellers, +; then this setting only turns them on and off; numeric settings will +; be treated as "yes". There are no special settings required for +; hardware echo cancellers; when present and enabled in their kernel +; modules, they take precedence over the software echo canceller compiled +; into DAHDI automatically. +; +echocancel=yes +; +; Generally, it is not necessary (and in fact undesirable) to echo cancel when +; the circuit path is entirely TDM. You may, however, change 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) +; +; WARNING: In some cases this option can make echo worse! If you are +; trying to debug an echo problem, it is worth checking to see if your echo +; is better with the option set to yes or no. Use whatever setting gives +; the best results. +; +; Note that these parameters do not apply to hardware echo cancellers. +; +;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 dialling *8#. For simple offices, just +; make these both the same. Groups range from 0 to 63. +; +callgroup=1 +pickupgroup=1 + +; +; Specify whether the channel should be answered immediately or if the simple +; switch should provide dialtone, read digits, etc. +; Note: If immediate=yes the dialplan execution will always start at extension +; 's' priority 1 regardless of the dialed number! +; +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 +; +; 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 +; +; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel +; basis if you would like that channel to behave like an SMDI message desk. +; The SMDI port specified should have already been defined in smdi.conf. The +; default port is /dev/ttyS0. +; +;usesmdi=yes +;smdiport=/dev/ttyS0 +; +; 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 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 if there was a +; hangup before the line was answered. This value can be tweaked to shorten +; how long it takes before DAHDI 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 +; +; This option specifies a preference for which music on hold class this channel +; should listen to when put on hold if the music class has not been set on the +; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer +; channel putting this one on hold did not suggest a music class. +; +; If this option is set to "passthrough", then the hold message will always be +; passed through as signalling instead of generating hold music locally. This +; setting is only valid when used on a channel that uses digital signalling. +; +; This option may be specified globally, or on a per-user or per-peer basis. +; +;mohinterpret=default +; +; This option specifies which music on hold class to suggest to the peer channel +; when this channel places the peer on hold. It may be specified globally or on +; a per-user or per-peer basis. +; +;mohsuggest=default +; +; PRI channels can have an idle extension and a minunused number. So long as +; at least "minunused" channels are idle, chan_dahdi 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 DAHDI (each one is 20ms, default is 4) +; +;jitterbuffers=4 +; +;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- +; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a + ; DAHDI channel. Defaults to "no". An enabled jitterbuffer will + ; be used only if the sending side can create and the receiving + ; side can not accept jitter. The DAHDI channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive DAHDI side will always + ; be used if the sending side can create jitter. + +; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. + +; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usually sent from exotic devices + ; and programs. Defaults to 1000. + +; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a DAHDI + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmax-size) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. + +; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". +;----------------------------------------------------------------------------------- +; +; You can define your own custom ring cadences here. You can define up to 8 +; pairs. If the silence is negative, it indicates where the callerid spill is +; to be placed. Also, if you define any custom cadences, the default cadences +; will be turned off. +; +; Syntax is: cadence=ring,silence[,ring,silence[...]] +; +; These are the default cadences: +; +;cadence=125,125,2000,-4000 +;cadence=250,250,500,1000,250,250,500,-4000 +;cadence=125,125,125,125,125,-4000 +;cadence=1000,500,2500,-5000 +; +; Each channel consists of the channel number or range. It inherits the +; parameters that were specified above its declaration. +; +; For GR-303, CRV's are created like channels except they must start with the +; trunk group followed by a colon, e.g.: +; +; crv => 1:1 +; crv => 2:1-2,5-8 +; +; +;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 distinctive 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 + diff --git a/configs/extensions.ael.sample b/configs/extensions.ael.sample index 89436dc21..ab8cdd854 100644 --- a/configs/extensions.ael.sample +++ b/configs/extensions.ael.sample @@ -27,7 +27,7 @@ globals { TRUNK="Zap/g2"; // Trunk interface // // Note the 'g2' in the TRUNK variable above. It specifies which group (defined - // in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in + // in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in // the specified group. The four possible options are: // // g: select the lowest-numbered non-busy Zap channel diff --git a/configs/extensions.conf.sample b/configs/extensions.conf.sample index d1e7a8310..b7151f4c4 100644 --- a/configs/extensions.conf.sample +++ b/configs/extensions.conf.sample @@ -90,7 +90,7 @@ IAXINFO=guest ; IAXtel username/password TRUNK=Zap/G2 ; Trunk interface ; ; Note the 'G2' in the TRUNK variable above. It specifies which group (defined -; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in +; in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in ; the specified group. The four possible options are: ; ; g: select the lowest-numbered non-busy Zap channel diff --git a/configs/sla.conf.sample b/configs/sla.conf.sample index 2e5c88b00..fa768c513 100644 --- a/configs/sla.conf.sample +++ b/configs/sla.conf.sample @@ -29,7 +29,7 @@ ;autocontext=line1 ; This supports automatic generation of the dialplan entries ; if the autocontext option is used. Each trunk should have - ; a unique context name. Then, in zapata.conf, this device + ; a unique context name. Then, in chan_dahdi.conf, this device ; should be configured to have incoming calls go to this context. ;ringtimeout=30 ; Set how long to allow this trunk to ring on an inbound call before hanging diff --git a/configs/smdi.conf.sample b/configs/smdi.conf.sample index 669530e81..62464c47b 100644 --- a/configs/smdi.conf.sample +++ b/configs/smdi.conf.sample @@ -2,7 +2,7 @@ [interfaces] ; Specify serial ports to listen for SMDI messages on below. These will be -; referenced later in zapata.conf. If you do not specify any interfaces then +; referenced later in chan_dahdi.conf. If you do not specify any interfaces then ; SMDI will be disabled. Interfaces can have several different attributes ; associated with them. diff --git a/configs/zapata.conf.sample b/configs/zapata.conf.sample deleted file mode 100644 index 21a2d6000..000000000 --- a/configs/zapata.conf.sample +++ /dev/null @@ -1,675 +0,0 @@ -; -; Zapata telephony interface -; -; Configuration file -; -; You need to restart Asterisk to re-configure the Zap channel -; CLI> reload chan_zap.so -; will reload the configuration file, -; but not all configuration options are -; re-configured during a reload. - - - -[trunkgroups] -; -; Trunk groups are used for NFAS or GR-303 connections. -; -; Group: Defines a trunk group. -; trunkgroup => ,[,...] -; -; trunkgroup is the numerical trunk group to create -; dchannel is the zap channel which will have the -; d-channel for the trunk. -; backup1 is an optional list of backup d-channels. -; -;trunkgroup => 1,24,48 -;trunkgroup => 1,24 -; -; Spanmap: Associates a span with a trunk group -; spanmap => ,[,] -; -; zapspan is the zap span number to associate -; trunkgroup is the trunkgroup (specified above) for the mapping -; logicalspan is the logical span number within the trunk group to use. -; if unspecified, no logical span number is used. -; -;spanmap => 1,1,1 -;spanmap => 2,1,2 -;spanmap => 3,1,3 -;spanmap => 4,1,4 - -[channels] -; -; Default language -; -;language=en -; -; Default context -; -context=default -; -; Switchtype: Only used for PRI. -; -; national: National ISDN 2 (default) -; dms100: Nortel DMS100 -; 4ess: AT&T 4ESS -; 5ess: Lucent 5ESS -; euroisdn: EuroISDN (also known as ETSI NET/5; Cisco calls this "primary-net5") -; ni1: Old National ISDN 1 -; qsig: Q.SIG -; -switchtype=national -; -; Some switches (AT&T especially) require network specific facility IE -; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet' -; -;nsf=none -; -; PRI Dialplan: Only RARELY used for PRI. -; -; unknown: Unknown -; private: Private ISDN -; local: Local ISDN -; national: National ISDN -; international: International ISDN -; dynamic: Dynamically selects the appropriate dialplan -; -;pridialplan=national -; -; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan) -; -; unknown: Unknown -; private: Private ISDN -; local: Local ISDN -; national: National ISDN -; international: International ISDN -; dynamic: Dynamically selects the appropriate dialplan -; -;prilocaldialplan=national -; -; PRI callerid prefixes based on the given TON/NPI (dialplan) -; This is especially needed for euroisdn E1-PRIs -; -; sample 1 for Germany -;internationalprefix = 00 -;nationalprefix = 0 -;localprefix = 0711 -;privateprefix = 07115678 -;unknownprefix = -; -; sample 2 for Germany -;internationalprefix = + -;nationalprefix = +49 -;localprefix = +49711 -;privateprefix = +497115678 -;unknownprefix = -; -; PRI resetinterval: sets the time in seconds between restart of unused -; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like -; channel restarts. so set the interval to a very long interval e.g. 100000000 -; or 'never' to disable *entirely*. -; -;resetinterval = 3600 -; -; Overlap dialing mode (sending overlap digits) -; -;overlapdial=yes -; -; Allow inband audio (progress) when a call is RELEASEd by the far end of a PRI -; -;inbandrelease=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). -; K: Layer 2 max number of outstanding unacknowledged I frames (default 7) -; N200: Layer 2 max number of retransmissions of a frame (default 3) -; T200: Layer 2 max time before retransmission of a frame (default 1000 ms) -; T203: Layer 2 max time without frames being exchanged (default 10000 ms) -; T305: Wait for DISCONNECT acknowledge (default 30000 ms) -; T308: Wait for RELEASE acknowledge (default 4000 ms) -; T309: Maintain active calls on Layer 2 disconnection (default -1, Asterisk clears calls) -; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s -; May vary in other ISDN standards (Q.931 1993 : 90000 ms) -; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms) -; -; 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)) -; fgccama Feature Group C-CAMA (DP DNIS, MF ANI) -; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI) -; 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 -; -; 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_ls -; -; If you have an outbound signalling format that is different from format -; specified above (but compatible), you can specify outbound signalling format, -; (see below). The 'signalling' format specified will be the inbound signalling -; format. If you only specify 'signalling', then it will be the format for -; both inbound and outbound. -; -; signalling=featdmf -; outsignalling=featb -; -; For Feature Group D Tandem access, to set the default CIC and OZZ use these -; parameters: -;defaultozz=0000 -;defaultcic=303 -; -; 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 milliseconds) -;toneduration=100 -; -; Whether or not to do distinctive ring detection on FXO lines -; -;usedistinctiveringdetection=yes -;distinctiveringaftercid=yes ; enable dring detection after callerid for those countries like Australia - ; where the ring cadence is changed *after* the callerid spill. -; -; 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 -; v23_jp = v23 as used in Japan -; dtmf = DTMF as used in Denmark, Sweden and Netherlands -; smdi = Use SMDI for callerid. Requires SMDI to be enabled (usesmdi). -; -;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 internal extensions -; With this set to 'yes', busy extensions will hear the call-waiting -; tone, and can use hook-flash to switch between callers. The Dial() -; app will not return the "BUSY" result for extensions. -; -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. -; See doc/callingpres.txt -; -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 -; -; For FXS ports (either direct analog or over T1/E1): -; Support flash-hook call transfer (requires three way calling) -; Also enables call parking (overrides the 'canpark' parameter) -; -; For digital ports using ISDN PRI protocols: -; Support switch-side transfer (called 2BCT, RLT or other names) -; This setting must be enabled on both ports involved, and the -; 'facilityenable' setting must also be enabled to allow sending -; the transfer to the ISDN switch, since it sent in a FACILITY -; message. -; -transfer=yes -; -; Allow call parking -; ('canpark=no' is overridden by 'transfer=yes') -; -canpark=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 without a voicemail -; context, then when voicemail is received in a mailbox in the default -; voicemail context in voicemail.conf, taking the phone off hook will cause a -; stutter dialtone instead of a normal one. -; -; If a mailbox is specified *with* a voicemail context, the same will result -; if voicemail received in mailbox in the specified voicemail context. -; -; for default voicemail context, the example below is fine: -; -;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. -; -; Note that when setting the number of taps, the number 256 does not translate -; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms. -; -; Note that if any of your Zaptel cards have hardware echo cancellers, -; then this setting only turns them on and off; numeric settings will -; be treated as "yes". There are no special settings required for -; hardware echo cancellers; when present and enabled in their kernel -; modules, they take precedence over the software echo canceller compiled -; into Zaptel automatically. -; -echocancel=yes -; -; Generally, it is not necessary (and in fact undesirable) to echo cancel when -; the circuit path is entirely TDM. You may, however, change 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) -; -; WARNING: In some cases this option can make echo worse! If you are -; trying to debug an echo problem, it is worth checking to see if your echo -; is better with the option set to yes or no. Use whatever setting gives -; the best results. -; -; Note that these parameters do not apply to hardware echo cancellers. -; -;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 dialling *8#. For simple offices, just -; make these both the same. Groups range from 0 to 63. -; -callgroup=1 -pickupgroup=1 - -; -; Specify whether the channel should be answered immediately or if the simple -; switch should provide dialtone, read digits, etc. -; Note: If immediate=yes the dialplan execution will always start at extension -; 's' priority 1 regardless of the dialed number! -; -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 -; -; 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 -; -; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel -; basis if you would like that channel to behave like an SMDI message desk. -; The SMDI port specified should have already been defined in smdi.conf. The -; default port is /dev/ttyS0. -; -;usesmdi=yes -;smdiport=/dev/ttyS0 -; -; 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 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 if 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 -; -; This option specifies a preference for which music on hold class this channel -; should listen to when put on hold if the music class has not been set on the -; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer -; channel putting this one on hold did not suggest a music class. -; -; If this option is set to "passthrough", then the hold message will always be -; passed through as signalling instead of generating hold music locally. This -; setting is only valid when used on a channel that uses digital signalling. -; -; This option may be specified globally, or on a per-user or per-peer basis. -; -;mohinterpret=default -; -; This option specifies which music on hold class to suggest to the peer channel -; when this channel places the peer on hold. It may be specified globally or on -; a per-user or per-peer basis. -; -;mohsuggest=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 -; -;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- -; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a - ; ZAP channel. Defaults to "no". An enabled jitterbuffer will - ; be used only if the sending side can create and the receiving - ; side can not accept jitter. The ZAP channel can't accept jitter, - ; thus an enabled jitterbuffer on the receive ZAP side will always - ; be used if the sending side can create jitter. - -; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. - -; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is - ; resynchronized. Useful to improve the quality of the voice, with - ; big jumps in/broken timestamps, usually sent from exotic devices - ; and programs. Defaults to 1000. - -; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a ZAP - ; channel. Two implementations are currently available - "fixed" - ; (with size always equals to jbmax-size) and "adaptive" (with - ; variable size, actually the new jb of IAX2). Defaults to fixed. - -; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". -;----------------------------------------------------------------------------------- -; -; You can define your own custom ring cadences here. You can define up to 8 -; pairs. If the silence is negative, it indicates where the callerid spill is -; to be placed. Also, if you define any custom cadences, the default cadences -; will be turned off. -; -; Syntax is: cadence=ring,silence[,ring,silence[...]] -; -; These are the default cadences: -; -;cadence=125,125,2000,-4000 -;cadence=250,250,500,1000,250,250,500,-4000 -;cadence=125,125,125,125,125,-4000 -;cadence=1000,500,2500,-5000 -; -; Each channel consists of the channel number or range. It inherits the -; parameters that were specified above its declaration. -; -; For GR-303, CRV's are created like channels except they must start with the -; trunk group followed by a colon, e.g.: -; -; crv => 1:1 -; crv => 2:1-2,5-8 -; -; -;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 distinctive 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 - -- cgit v1.2.3