[asterisk-ss7] ansi clarification

Tom Chandler tchandle at eastex.net
Sat Jun 14 18:56:13 CDT 2008

listed below per request is zapata.conf

/tom c


; DAHDI telephony


; Configuration file


; You need to restart Asterisk to re-configure the DAHDI channel

; CLI> reload chan_dahdi.so

; will reload the configuration file,

; but not all configuration options are

; re-configured during a reload (signalling, as well as

; PRI and SS7-related settings cannot be changed on a

; reload.


; This file documents many configuration variables. Normally unless you

; know what a variable means or that it should be changed, there's no

; reason to unrem lines.


; remmed-out examples below (those lines that begin with a ';' but no

; space afterwards) typically show a value that is not the defauult value,

; but would make sense under cetain circumstances. The default values

; are usually sane. Thus you should typically not touch them unless you

; know what they mean or you know you should change them.



; Trunk groups are used for NFAS or GR-303 connections.


; Group: Defines a trunk group.

; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]


; 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 => 1,23,24

;trunkgroup => 1,24


; Spanmap: Associates a span with a trunk group

; spanmap => <dahdispan>,<trunkgroup>[,<logicalspan>]


; 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



; Default language




; Context for calls. Defaults to 'default'




; Switchtype: Only used for PRI.


; national: National ISDN 2 (default)

; dms100: Nortel DMS100

; 4ess: AT&T 4ESS

; 5ess: Lucent 5ESS

; euroisdn: EuroISDN (common in Europe)

; ni1: Old National ISDN 1

; qsig: Q.SIG




; Some switches (AT&T especially) require network specific facility IE

; supported values are currently 'none', 'sdn', 'megacom', 
'tollfreemegacom', 'accunet'


; nsf cannot be changed on a reload.




; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used 

; the dialed number. For most installations, leaving this as 'unknown' (the

; default) works in the most cases. In some very unusual circumstances, you

; may need to set this to 'dynamic' or 'redundant'. Note that if you set one

; of the others, you will be unable to dial another class of numbers. For

; example, if you set 'national', you will be unable to dial local or

; international numbers.


; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's

; numbering plan). In North America, the typical use is sending the 10 digit

; callerID number and setting the prilocaldialplan to 'national' (the 

; Only VERY rarely will you need to change this.


; Neither pridialplan nor prilocaldialplan can be changed on reload.


; unknown: Unknown

; private: Private ISDN

; local: Local ISDN

; national: National ISDN

; international: International ISDN

; dynamic: Dynamically selects the appropriate dialplan

; redundant: Same as dynamic, except that the underlying number is not

; changed (not common)





; pridialplan may be also set at dialtime, by prefixing the dialled number 

; one of the following letters:

; U - Unknown

; I - International

; N - National

; L - Local (Net Specific)

; S - Subscriber

; V - Abbreviated

; R - Reserved (should probably never be used but is included for 


; Additionally, you may also set the following NPI bits (also by prefixing 

; dialled string with one of the following letters):

; u - Unknown

; e - E.163/E.164 (ISDN/telephony)

; x - X.121 (Data)

; f - F.69 (Telex)

; n - National

; p - Private

; r - Reserved (should probably never be used but is included for 


; You may also set the prilocaldialplan in the same way, but by prefixing 

; Caller*ID Number, rather than the dialled number. Please note that telcos

; which require this kind of additional manipulation of the TON/NPI are 

; Most telco PRIs will work fine simply by setting pridialplan to unknown or

; dynamic.



; PRI caller ID prefixes based on the given TON/NPI (dialplan)

; This is especially needed for EuroISDN E1-PRIs


; None of the prefix settings can be changed on reload.


; 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

; B channels; defaults to 'never'.


;resetinterval = 3600


; Overlap dialing mode (sending overlap digits)

; Cannot be changed on a reload.




; 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 (default)


; priindication cannot be changed on a reload.


;priindication = outofband


; If you need to override the existing channels selection routine and force 

; PRI channels to be marked as exclusively selected, set this to yes.


; priexclusive cannot be changed on a reload.


;priexclusive = yes


; ISDN Timers

; All of the ISDN timers and counters that are used are configurable. 

; 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.

; Cannot be changed on a reload.


;facilityenable = yes


; pritimer cannot be changed on a reload.


; Signalling method. The default is "auto". Valid values:

; auto: Use the current value from DAHDI.

; em: E & M

; em_e1: E & M E1

; 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

; ss7: Signalling System 7


; 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)

; ss7: Signalling System 7


; signalling of a channel can not be changed on a reload.




; If you have an outbound signalling format that is different from format

; specified above (but compatible), you can specify outbound signalling 

; (see below). The 'signalling' format specified will be the inbound 

; format. If you only specify 'signalling', then it will be the format for

; both inbound and outbound.


; outsignalling can only be one of:

; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,

; featdmf, featdmf_ta, e911, fgccama, fgccamamf


; outsignalling cannot be changed on a reload.






; For Feature Group D Tandem access, to set the default CIC and OZZ use 

; parameters (Will not be updated on reload):





; A variety of timing parameters can be specified as well

; The default values for those are "-1", which is to use the

; compile-time defaults of the DAHDI kernel modules. The timing

; parameters, (with the standard default from DAHDI):


; 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)


; None of them will update on a reload.


; How long generated tones (DTMF and MF) will be played on the channel

; (in milliseconds).


; This is a global, rather than a per-channel setting. It will not be

; updated on a reload.




; Whether or not to do distinctive ring detection on FXO lines:




; enable dring detection after caller ID for those countries like Australia

; where the ring cadence is changed *after* the caller ID spill:




; Whether or not to use caller ID:




; Hide the name part and leave just the number part of the caller ID

; string. Only applies to PRI channels.



; Type of caller ID signalling in use

; bell = bell202 as used in US (default)

; 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 caller ID. Requires SMDI to be enabled (usesmdi).




; What signals the start of caller ID

; ring = a ring signals the start (default)

; polarity = polarity reversal signals the start

; polarity_IN = polarity reversal signals the start, for India,

; for dtmf dialtone detection; using DTMF.

; (see doc/India-CID.txt)




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

; (If your dialplan doesn't catch it)




; The following option enables receiving MWI on FXO lines. The default

; value is no. When this is enabled, and MWI notification indicates on or 

; the script specified by the mwimonitornotify option is executed. Also, an

; internal Asterisk MWI event will be generated so that any other part of

; Asterisk that cares about MWI state changes will get notified, just as if

; the state change came from app_voicemail. The energy level that must be 

; before starting the MWI detection process can be set with 'mwilevel'.





; This option is used in conjunction with mwimonitor. This will get executed

; when incoming MWI state changes. The script is passed 2 arguments. The

; first is the corresponding mailbox, and the second is 1 or 0, indicating 

; there are messages waiting or not.




; 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.




; 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 

; calling switch is sending.

; See README.callingpres. FIXME: file no longer exists.




; Some countries (UK) have ring tones with different ring tones (ring-ring),

; which means the caller ID needs to be set later on, and not just after

; the first ring, as per the default (1).


;sendcalleridafter = 2



; Support caller ID on Call Waiting




; Support three-way calling




; 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.




; Allow call parking

; ('canpark=no' is overridden by 'transfer=yes')




; Support call forward variable




; Whether or not to support Call Return (*69, if your dialplan doesn't

; catch this first)




; 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 

; 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:




; for any other voicemail context, the following will produce the stutter 


;mailbox=1234 at 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 

; 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.





; As of Zaptel 1.4.8, some DAHDI echo cancellers (software and hardware)

; support adjustable parameters; these parameters can be supplied as

; additional options to the 'echocancel' setting. Note that Asterisk

; does not attempt to validate the parameters or their values, so if you

; supply an invalid parameter you will not know the specific reason it

; failed without checking the kernel message log for the error(s)

; put there by DAHDI.




; Generally, it is not necessary (and in fact undesirable) to echo cancel 

; the circuit path is entirely TDM. You may, however, change this behavior

; by enabling the echo canceller 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

; DAHDI 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.





; 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)


; Gain Settings: increasing / decreasing the volume level on a channel.

; The values are in db (decibells). A positive number

; increases the volume level on a channel, and a

; negavive value decreases volume level.


; There are several independent gain settings:

; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0

; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel.

; Default: 0.0

; cid_rxgain: set the gain just for the caller ID sounds Asterisk

; emits. Default: 5.0 .




; Logical groups can be assigned to allow outgoing roll-over. Groups range

; from 0 to 63, and multiple groups can be specified. By default the

; channel is not a member of any group.


; Note that an explicit empty value for 'group' is invalid, and will not

; override a previous non-empty one. The same applies to callgroup and

; pickupgroup as well.




; 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. Groups range from 0 to 63.




; Channel variable to be set for all calls from this channel



; Specify whether the channel should be answered immediately or if the 

; switch should provide dialtone, read digits, etc.

; Note: If immediate=yes the dialplan execution will always start at 

; 's' priority 1 regardless of the dialed number!




; Specify whether flash-hook transfers to 'busy' channels should complete or

; return to the caller performing the transfer (default is yes).




; caller ID can be set to "asreceived" or a specific number if you want to

; override it. Note that "asreceived" only applies to trunk interfaces.

; fullname sets just the


; fullname: sets just the name part.

; cid_number: sets just the number part:


;callerid = 123456


;callerid = My Name <2564286000>

; Which can also be written as:

;cid_number = 2564286000

;fullname = My Name


;callerid = asreceived


; should we use the caller ID from incoming call on DAHDI transfer?


;useincomingcalleridondahditransfer = yes


; 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

; billing




; ADSI (Analog Display Services Interface) can be enabled on a per-channel

; basis if you have (or may have) ADSI compatible CPE equipment




; 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.





; 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.




; If busydetect is enabled, it is also possible to specify how many busy 

; to wait for before hanging up. The default is 3, but it might be

; safer to 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.




; If busydetect is enabled, it is also possible to specify the cadence of 

; busy signal. In many countries, it is 500msec on, 500msec off. Without

; busypattern specified, we'll accept any regular sound-silence pattern that

; repeats <busycount> times as a busy signal. If you specify busypattern,

; then we'll further check the length of the sound (tone) and silence, which

; will further reduce the chance of a false positive.




; NOTE: In make menuselect, 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 enabling the



; To further detect which hangup tone your telco provider is sending, it is

; useful to use the ztmonitor utility to record the audio that main/dsp.c

; is receiving after the caller hangs up.


; Use a polarity reversal to mark when a outgoing call is answered by the

; remote party.




; In some countries, a polarity reversal is used to signal the disconnect of 

; phone line. If the hanguponpolarityswitch option is selected, the call 

; be considered "hung up" on a polarity reversal.




; polarityonanswerdelay: minimal time period (ms) between the answer

; polarity switch and hangup polarity switch.

; (default: 600ms)


; 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.


; Few zones are supported at the time of this writing, but may be selected

; with "progzone".


; progzone also affects the pattern used for buzydetect (unless

; busypattern is set explicitly). The possible values are:

; us (default)

; ca (alias for 'us')

; cr (Costa Rica)

; br (Brazil, alias for 'cr')

; uk


; This feature can also easily detect false hangups. The symptoms of this is

; being disconnected in the middle of a call for no reason.





; Set the tonezone. Equivalent of the defaultzone settings in

; /etc/dahdi.conf . This sets the tone zone by number.

; Note that you'd still need to load tonezones (loadzone in dahdi.conf).

; The default is -1: not to set anything.

;tonezone = 0 ; 0 is US


; FXO (FXS signalled) devices must have a timeout to determine if there was 

; 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 


; ringtimeout will not update on a reload.




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

; Pulse digits from phones (FXS devices, FXO signalling) are always

; detected.




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







; This option specifies a preference for which music on hold class this 

; should listen to when put on hold if the music class has not been set on 

; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the 

; 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 

; passed through as signalling instead of generating hold music locally. 

; setting is only valid when used on a channel that uses digital signalling.




; This option specifies which music on hold class to suggest to the peer 

; when this channel places the peer on hold.




; 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 

; 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 

; multilink PPP, thus more efficiently utilizing combined voice/data 

; than conventional fixed mappings/muxings.


; Those settings cannot be changed on reload.



;idleext=6999 at dialout




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

; This is set globally, rather than per-channel.




;------------------------------ 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 caller ID spill 

; to be placed. Also, if you define any custom cadences, the default 

; will be turned off.


; This setting is global, rather than per-channel. It will not update on

; a reload.


; Syntax is: cadence=ring,silence[,ring,silence[...]]


; These are the default cadences:







; 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 

; 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" <(630) 372-1564>

;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 cannot be changed on a reload.


; 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 

; and they will be printed on the console when an inbound call comes in.


; dringXrange is used to change the acceptable ranges for "tone offsets". 
Defaults to 10.

; Note: a range of 0 is NOT what you might expect - it instead forces it to 
the default.

; A range of -1 will force it to always match.

; Anything lower than -1 would presumably cause it to never match.








; If no pattern is matched here is where we go.


;channel => 1

; ---------------- Options for use with signalling=ss7 -----------------

; None of them can be changed by a reload.


; Variant of SS7 signalling:

; Options are itu and ansi

ss7type = ansi

; SS7 Called Nature of Address Indicator


; unknown: Unknown

; subscriber: Subscriber

; national: National

; international: International

; dynamic: Dynamically selects the appropriate dialplan




; SS7 Calling Nature of Address Indicator


; unknown: Unknown

; subscriber: Subscriber

; national: National

; international: International

; dynamic: Dynamically selects the appropriate dialplan





; sample 1 for Germany

;ss7_internationalprefix = 00

;ss7_nationalprefix = 0

;ss7_subscriberprefix =

;ss7_unknownprefix =


; All settings apply to linkset 1

linkset = 1

; Point code of the linkset. For ITU, this is the decimal number

; format of the point code. For ANSI, this can either be in decimal

; number format or in the xxx-xxx-xxx format

pointcode = 001-001-001

; Point code of node adjacent to this signalling link (Possibly the STP 
between you and

; your destination). Point code format follows the same rules as above.

adjpointcode = 248-xxx-xxx

adjpointcode = 248-xxx-xxx

; Default point code that you would like to assign to outgoing messages (in 
case of

; routing through STPs, or using A links). Point code format follows the 
same rules

; as above.

defaultdpc = 001-001-002

; Begin CIC (Circuit indication codes) count with this number

cicbeginswith = 1

; What the MTP3 network indicator bits should be set to. Choices are

; national, national_spare, international, international_spare


; First signalling channel

sigchan = 23

sigchan = 24

; Channels to associate with CICs on this linkset

channel = 1-22


; For more information on setting up SS7, see the README file in libss7 or

; the doc/ss7.txt file in the Asterisk source tree.

; ----------------- SS7 Options ----------------------------------------

; Configuration Sections

; ~~~~~~~~~~~~~~~~~~~~~~

; You can also configure channels in a separate dahdi.conf section. In

; this case the keyword 'channel' is not used. Instead the keyword

; 'dahdichan' is used (as in users.conf) - configuration is only processed

; in a section where the keyword dahdichan is used. It will only be

; processed in the end of the section. Thus the following section:



;echocancel = 64

;dahdichan = 1-8

;group = 1


; Is somewhat equivalent to the following snippet in the section

; [channels]:


;echocancel = 64

;group = 1

;channel => 1-8


; When starting a new section almost all of the configuration values are

; copied from their values at the end of the section [channels] in

; dahdi.conf and [general] in users.conf - one section's configuration

; does not affect another one's.


; Instead of letting common configuration values "slide through" you can

; use configuration templates to easily keep the common part in one

; place and override where needed.



;echocancel = yes

;group = 0,4

;callgroup = 3

;pickupgroup = 3

;threewaycalling = yes

;transfer = yes

;context = phones

;faxdetect = incoming



;dahdichan = 1

;callerid = My Name <501>

;mailbox = 501 at mailboxes




;dahdichan = 2

;faxdetect = no

;context = fax



;dahdichan = 3

;pickupgroup = 3,4

----- Original Message ----- 
From: "Matthew Fredrickson" <creslin at digium.com>
To: <asterisk-ss7 at lists.digium.com>
Sent: Saturday, June 14, 2008 5:40 PM
Subject: Re: [asterisk-ss7] ansi clarification

> Tom Chandler wrote:
>> After some testing, I am confused about how libss7 works in
>> an ANSI A-link setup
>> (all dummy point codes)
>> setup
>> Switch A
>> pointcode    =     001-001-001
>> adjpointcode  =    248-001-001        (STP )
>> adjpointcode  =    248-001-002        (STP )
>> defaultdpc =    001-001-002
>> Switch B
>> pointcode  =  001-001-002
>> adjpointcode (not defined)
>> defaultdpc = 001-001-001
>> My question is on linkset alignment, should the linkset use the 
>> adjpointcode to align
>> the links.  It appears that it is using the defaultdpc.  (That would be 
>> F-Link setup).
>> My linkset comes up and is aligned, and calls pass, which if a true 
>> A-Link setup, should
>> not happen......
>> I am confused that my links are aligning, and the 248 point codes do not 
>> exist.  If it is
>> a true A-link then the alignment should be to the adjointcodes ??
>> What am I missing or what is happening......
> Can you post your zapata.conf?
> -- 
> Matthew Fredrickson
> Software/Firmware Engineer
> Digium, Inc.
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7 

More information about the asterisk-ss7 mailing list