- This tutorial requires you to have read the general E1/T1 tutorial, and you to have a working and provisioned PRI link.
You should also know what signalling protocol and what framing protocol the link is using.
Since you are looking at the tutorial for PRI's, (and not for E&M or FXO/FXS) the signalling will always be CCS.
If you are using an E1, the framing could be HDB3 or AMI. (hdb3 is more common, if you are brute forcing the info, try this one first :)
If you are using a T1, the framing could be b8zs or AMI.
Before complaining that things don't work, also have a look at the tutorial link for you specific card. (you might have to set some jumpers on the card before starting).
1. Installation of the software
1.1. Libpri
1.2. Zaptel
Loading the modules:
1.3. Asterisk
2. Configuring asterisk and zaptel for the card:
2.1.: /etc/zaptel.conf
It is important to know what signalling type you will be using.
If you do not control the other end of the link, (e.g.: you are connecting to a carrier, or you are connecting to an unkown pbx) you need to ask the carrier, or the person responsable for that pbx what signalling and framing type you should use.
The format for zaptel.conf is very simple, each line is of the type:
parameter=value
When multiple values are needed, the values are separated by commas:
e.g.:
parameter=value1,value2,value3
lines starting with a hash symbol (#) are comments and will be ignored. (In fact, everything behind a # will be ignored.
e.g.:
parameter=value #explanation
In the case of PRI setups, the first (non comments) lines in zaptel.conf should be the span definitions for the E1/T1's.
A span definition is in the format:
span=<spannum>,<timing>,<LBO>,<framing>,<coding>
Before going any further, lets have a closer look at these options:
spannum:
This defines the number of the span (=port). This is counted across the cards. (have a look at the picture of the te4xxp cards to see where it starts counting.
E.g.:
If you have a quad e1/t1 card, and a single port e1/t1 card, you will need 5 spans.
Span 1 might be on the first card or on the second card, so there are two options:
span 1 = first port on the single port card.
span 2,3,4,5 = the ports on the quad port card.
or span 1,2,3,4 = the ports on the quad port card
span 5 = the first port on the single port card.
framing and coding:
Lets have a look at the possible options for a T1 setup:
- Framing:
D4 or ESF
- Coding:
AMI or B8ZS
Usually when the line is ESF framing, the line coding is B8ZS and when the line is D4 SF framing, the line coding is usually AMI.
e.g.: span=1,1,0,esf,b8zs
Lets have a look at the possible options for an E1 setup:
- Framing:
CCS or CAS
- Coding:
AMI or HDB3
On an E1 card, you can optionally enable CRC checking.
e.g.: e.g.: span=1,1,0,CCS,HDB3,crc
Timing:
As mentioned before, every line needs to have a synching time source on one of both ends. (An asterisk PRI line can be clocked internally or can be clocked by the telco or the PBX it is connected to.)
Lets rephrase this to make this a little more clear:
Internal timing is when the timing is taken from the clock in the card.
External timing is when the timing is taken from the line on the PRI.
Timing is done PER card, so every card needs its own timing sources defined. (they can NOT be shared across cards.)
The timing parameter in the span definition in zaptel.conf determines the selection of primary, secondary, and so on sync sources. If the span you are defining should be considered a primary sync source, then give it a value of "1".
(This means, zaptel will take the timing as received on the line connected to this port (span). This is called external clocking. if no timing was received on this line, it will fallback to internal clocking, meaning it will look at its own clock chip and generate timing,)
To define a secondary clock source, use "2", to define a 3rd one, use 3, and so on.
In this case, zaptel will look if any timing is provided by the carrier, on the line connected to span 1, if its found it will use this timing for the complete card. If no timing is found it will have a look if a timing is found on span 2, if it finds a timing on the line, it will use this one, if it does not, it will use its internal clock for the timing.
the zero in the line for the 3rd span means that zaptel will never use the timing provided by the carrier or pbx connected to the 3rd span, it will disregard it if found.
Another definition for this might be:
0 means internal clocking
1 means recovered from this span
2 means recovered from this span if the span with clock=1 is down
The term loop is sometimes also used for external timing taken from the carrier.
Line Build Out (LBO)
The Line Build Out (dB) parameter is a value for a power level that is set based on the distance from the card to the T-1/E-1 service provider's gateway. If the CSU/DSU card is close by, the gateway requires less power and the line build out value is lower; if the card is far away, the gateway requires more power and the line build out value is higher. This setting is determined by the T-1/E-1 service provider. Valid options are:
Note the yellow parameter i added to some of the lines. When setting this parameter, a yellow alarm will be set on this pri when no channels are open. (e.g. when zaptel is loaded, but asterisk is not yet running, the the remote pri will know no channels are open and might send it to another pri, without having to try first).
Channel configurations
Now that the spans are configured, lets have a look at the rest of the configuration file:
Next up is the configuration of the channels. as seen before a PRI has B channels and D channels.
In the case of a T1 PRI, the typical setup will be:
bchan=1-23
dchan=24
In the case of an E1 PRI, the typical setup will be:
bchan=1-15,17-31
dchan=16
But this is only for one span, and if the channels on the first PRI port happen to be the first channels found by zaptel in your system.
If for example you would also have an TDM card or an FXO card, then channel 1 might be for these cards and the channels for the pri would start counting at channel 2.
In that case, the configurationf for the first PRI would look like this:
In the case of a T1 PRI:
bchan=2-24
dchan=25
In the case of an E1 PRI:
bchan=2-16,18-32
dchan=17
In case you have multiple E1/T1 lines, you just need to add multiple bchan and dchan lines.
E.g. a four port E1 configuration could have these lines:
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109
Tonezones
Now that we showed you have to define the channels, the only thing left to explain is the tonezones.
Two options can be set, the tonezone=<zone> option and the defaulttonezone=<zone>, where zone is a two letter country code.
By setting these options, it is possible to preload one or more tone zones.
This will prevent them from getting overwritten by other users (if you allow non-root users to open /dev/tor* interfaces anyway. Also this means they won't have to be loaded at runtime.
Editorial note: this is taken from an old zaptel.conf, i don't know if the overwriting prevention also applies to the newer /dev/zap/*.
If you know, let us know by posting a comment. (i dont feel like browsing code for this one.)
The tones zones are to allow channels used by British users to be configured to give familiar UK tones, while a dutch user could hear hear French tones on another channel.
These indications are for example:
DIALTONE
BUSY
RINGTONE
CONGESTION
CALLWAIT
DIALRECALL
RECORDTONE
INFO
STUTTER
These tone zones will be used for users connected to the PRI leg of the call, (e.g. incoming calls on the PRI) as well as outgoing calls (coming from e.g. SIP).
They will only work if the carrier is not sending audio tones for these indications.
Currently only a few tone sets are supported, do a:
cat /usr/src/zaptel/zonedata.c | grep ",
on your system to see which ones.
some possibilites listed here:
us: United States / North America
au: Australia
fr: France
nl: Netherlands,
uk: United Kingdom
fi: Finland
es: Spain
jp: Japan,
no: Norway
at: Austria
nz: New Zealand,
it: Italy
us-old United States Circa 1950/ North America
gr: Greece
tw: Taiwan
cl: Chile
se: Sweden
be: Belgium
sg: Singapore
il: Israel
br: Brazil
hu: Hungary
lt: Lithuania
pl: Poland
za: South Africa
pt: Portugal
ee: Estonia
mx: Mexico
in: India
de: Germany
ch: Switzerland
dk: Denmark
cz: Czech Republic
cn: China
This could be a valid configuration:
loadzone=us,nl,it,cn : preload the tones for these countries
defaultzone=it: means use the tones for this zone by default (when no other zone is specified).
2.2.: /etc/asterisk/zapata.conf
Lets have a look at how the zapata.conf file is configured:
The file consists of a bunch of values with the form option=parameter or channel => numbers separated by comma's.
Comments are done by putting a semicolon in front of the comments.
e.g.:
; this is a comment only line and will be ignored
channel => 1 ; comments here, everything after the ; will be ignored.
The first not commented line should be [channels].
The format and behaviour of zapata.conf is not the same as the other asterisk configuration files. (to be honest, it s*cks bigtime!!!!).
Every time a channel => keyword is used, asterisk will look create this channel or these channels, with as parameters whatever was filled on for the keywords above.
This probably sounds a little too complex, so we will explain this with a small example:
We just created two channels. 1 and 2, the first one will have parameters, context=reception (and not context=blabla), signalling=fxo_ks, language-en.
The second one will have signalling=fxo_ks, language=fr but will also have context=reception as the keyword reception was assigned a value before.
I'm sure you agree that this can lead to a lot of unexpected behaviour.
Now that we have this covered, lets see what we can define for pri cards:
on top of the zapata.conf file, you can specify the default language and default context, these will be overwritten with any other context=line or language=line. (whatever line was typed last will be the new default - yes, it is as stupid as it sounds...).
switchtype=
The switchtype defines what kind of isdn dialect the pri is speaking, the telco should be able to tell you this. In europe, everyone uses the same standard, euroisdn.
Options for T1:
national: National ISDN 2 (default)
dms100: Nortel DMS100
4ess: AT&T 4ESS
5ess: Lucent 5ESS
ni1: Old National ISDN 1
Options for E1:
euroisdn: EuroISDN
pridialplan=
unknown: Unknown
private: Private ISDN
local: Local ISDN
national: National ISDN
international: International ISDN
dynamic: Only works on cvs-head or asterisk 1.2 and above.
overlapdial= yes or no
signalling=
Since we are using PRI's, the only two options here are:
pri_net
pri_cpe
;
; 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.
; group => <trunkgroup>,<dchannel>[,<backup1>...]
;
; 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>,<trunkgroup>[,<logicalspan>]
;
; 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
; 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', 'accunet'
;
;nsf=none
;
; PRI Dialplan: Only RARELY used for PRI.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
;
;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
;
;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
;
; 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
;
; 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)
;
; 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:
; pri_cpe: PRI signalling, CPE side
; pri_net: PRI signalling, Network side
signalling=fxo_ls
;
; How long generated tones (DTMF and MF) will be played on the channel (in miliseconds)
;toneduration=100
;
;
; 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, dtmf = DTMF as used in Denmark, Sweden and Netherlands
;
;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 restrict outgoing caller ID (will be sent as ANI only, not available for the user)
; Mostly use with FXS ports
;
;restrictcid=no
;
; Whether or not use the caller ID presentation for the outgoing call that the calling switch is sending
;
usecallingpres=yes
;
; 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
;
; Support flash-hook call transfer (requires three way calling)
; Also enables call parking (overrides the 'canpark' parameter)
;
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 recieved 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.
;
echocancel=yes
;
; Generally, it is not necessary (and in fact undesirable) to echo cancel
; when the circuit path is entirely TDM. You may, however, reverse this
; behavior by enabling the echo cancel during pure TDM bridging below.
;
echocancelwhenbridged=yes
;
; In some cases, the echo canceller doesn't train quickly enough and there
; is echo at the beginning of the call. Enabling echo training will cause
; asterisk to briefly mute the channel, send an impulse, and use the impulse
; response to pre-train the echo canceller so it can start out with a much
; closer idea of the actual echo. Value may be "yes", "no", or a number of
; milliseconds to delay before training (default = 400)
;
;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 dialing *8#. For simple offices, just
; make these both the same
;
callgroup=1
pickupgroup=1
;
; Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
;
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
;
; 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
;
; For fax detection, uncomment one of the following lines. The default is *OFF*
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;
; Select which class of music to use for music on hold. If not specified
; then the default will be used.
;
;musiconhold=default
;
; PRI channels can have an idle extension and a minunused number. So long
; as at least "minunused" channels are idle, chan_zap will try to call
; "idledial" on them, and then dump them into the PBX in the "idleext"
; extension (which is of the form exten@context). When channels are needed
; the "idle" calls are disconnected (so long as there are at least "minidle"
; calls still running, of course) to make more channels available. The
; primary use of this is to create a dynamic service, where idle channels
; are bundled through multilink PPP, thus more efficiently utilizing
; combined voice/data services than conventional fixed mappings/muxings.
;
;idledial=6999
;idleext=6999@dialout
;minunused=2
;minidle=1
;
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
;
;jitterbuffers=4
;
; 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
;
;
;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
;
; For example, maybe we have some other channels
; which start out in a different context and use
; E & M signalling instead.
;
;context=remote
;sigalling=em
;channel => 15
;channel => 16
;signalling=em_w
;
; All those in group 0 I'll use for outgoing calls
;
; Strip most significant digit (9) before sending
;
;stripmsd=1
;callerid=asreceived
;group=0
;signalling=fxs_ls
;channel => 45
;signalling=fxo_ls
;group=1
;callerid="Joe Schmoe" <(256) 428-6131>
;channel => 25
;callerid="Megan May" <(256) 428-6132>
;channel => 26
;callerid="Suzy Queue" <(256) 428-6233>
;channel => 27
;callerid="Larry Moe" <(256) 428-6234>
;channel => 28
;
; Sample PRI (CPE) config: Specify the switchtype, the signalling as
; either pri_cpe or pri_net for CPE or Network termination, and generally
; you will want to create a single "group" for all channels of the PRI.
;
; switchtype = national
; signalling = pri_cpe
; group = 2
; channel => 1-23
;
; Used for distintive ring support for x100p.
; You can see the dringX patterns is to set any one of the dringXcontext fields
; and they will be printed on the console when an inbound call comes in.
;
;dring1=95,0,0
;dring1context=internal1
;dring2=325,95,0
;dring2context=internal2
; If no pattern is matched here is where we go.
;context=default
;channel => 1
B) Add this to the bottom of /etc/asterisk/zapata.conf
signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=national
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds
callerid=asreceived
group=1
context=default ; Points to the default context of your extensions.conf
channel => 1-23 ; Set this to 1-15,17-31 for E1
C) Add this line to the end of the [default] context in your extensions.conf file
exten => _XXXXXX,1,Dial,Zap/g1/${EXTEN} ; Press any 7 digit number and try to dial that number through Zap channel 1
exten => s,1,Wait(1)
exten => s,2,Answer
exten => s,2,Playback(demo-congrats) ; Plays the demo-congrats file after answering the line
exten => s,3,Hangup
If you comment out overlapdial=yes (or set it to no) and the outgoing calls no longer work, this might be because you set
pridialplan= to something other than unknown
(e.g. when its set to local, which is supposed to be only for local calls).
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Cesar Anaya (cesar dot anayar at gmail dot com) 27 May 2017 08:48:43 Regarding the Line Build out Wrong DB parameters set up:
If the wrong one gets chosen, The service Provider will not receive a signal from the PBX and AIS alarms will be generated.
Beacuse the signal from the pbx will be either to strong or to weak to work.
PBX will need to re config the build out to fix this and adjust DB accordingly depending on the distance from the PBX to the SJ.
If PRI cards are replaced you might need to reconfig this settings!
-Cesar Anaya
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Praveen Rajan (praveenrajan at hotmail dot com) 23 July 2008 11:00:45 Hi,
For Airtel PRI the Conf. shold be:-
For Zaptel.conf:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
And for Zapata.conf:
group=0
signalling=pri_cpe
switchtype=euroisdn
channel=>1-15, 17-31
PL. check with the provider again.
Regards,
Praveen
Praveen Rajan (praveenrajan at hotmail dot com) 23 July 2008 11:00:21 Hi,
For Airtel PRI the Conf. shold be:-
For Zaptel.conf:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
And for Zapata.conf:
group=0
signalling=pri_cpe
switchtype=euroisdn
channel=>1-15, 17-31
PL. check with the provider again.
Regards,
Praveen
Ramon (rginez at nitido dot com) 03 August 2007 21:41:27 Hello i was wondering to know , if we can test e1 connections with the same card, i mean connect a cable from one port to another and set the asterisk for example: set g1 to route calls on p1, connect p1 with p2 with a cable, and set g2 with p2 and context="anycontext" and try to make calls from g1 to "anycontext"
dirk rowdies (support at microsoft dot com) 03 May 2007 21:57:51 hi we have successfully installed and tested workin digium,sangoma and rhino cards. please email us for more information if you need help with setup.
Mardonio Sanchez (mar2xs at homail dot com) 24 April 2007 23:01:50 I am using TE110P with X100P receiving a busy/congested message when using PRI line and with yellow/red alarm in the pri.
CLI> zap show status
Description Alarms IRQ bpviol CRC4
accountcode=Tw-analog
signalling=fxs_ks
group=1
channel => 25
Any advice?
Thanks.
mohankumar (kumar_mohin at sify dot com) 10 April 2007 11:07:40 Hi...I am using x100p card zaptel card and my zaptel and zapata configurations are
zapata.conf
[trunkgroups]
[channels]
context=unused
signalling=fxs_ks
channel => 1
zaptel.conf
loadzone=us
defaultzone=us
fxsks=1
But still getting the error when i restart the system...I have even loaded the modules.Pls help me to solve the isssue...
Fred (fred dot bonsu at soacom dot eu) 07 December 2006 22:29:21 I'm having problems configuring Asterisk. I have TE405P card and want to connect it to a GSM Gateway with PRI interface.
My first problem is, Asterisk gives the ff. error when started:
"Asterisk ended with exist status 1" Asterisk died with code 1". What does this mean?
One thing I have observed is when I on comment channel declaration in Zapata config then the error vanished. Could someone show me what to do?
My second problem is how I could get Asterisk to communicate with the GSM gateway. Is a way to configure Asterisk to communicate with the gateway? I have a Novatec (from Germany) TGM 20 gateway with PRI interface. Is someone out there familiar with this GMS gateway and successfully configured with Asterisk?
Thanks.
Raffaele (cozzella at yahoo dot com) 04 December 2006 18:42:46 Hi I would like to know if it is possible to configure Asterisk in this way: For each call make a Three way call with the third agent which answers and plays automatically a music
Thanks Raf
Vijay (vijay dot sinha4u at gmail dot com) 18 October 2006 08:49:31 I am using an Airtel PRI connected to my Sangoma A104 card. For the Zapata.conf file do you suggest the following are correct:
Remaining entries for this file I believe are correctly entered.
Anand Kumar Gupta (kumaranandgupta at gmail dot com) 22 September 2006 12:33:31 Document quality is very good.Please give a example of the following in the docume i.e. 1.)We are the client of a PSTN provider,2.)One port T1/E1 card and using on E1 line.3.)If we want to use only one line(not through a group) then exten => _xxxx.,1,Dial(??????)
Abdullah Al-Fukr (myjihad at terror dot org) 25 July 2006 11:12:39 plz send me instructions on how to build asterix PABX for free, quickly because I need it up and running now. send clear instructions and all details so I don't have to do the work. Hurry.
Mohammad Zeeshan (mzeeshanfahd at gmail dot com) 21 June 2006 14:53:04 Can u tell me that when connecting to the legacy PBX or PSTN line if the e1/t1 card does not come out of red alarm then is there any setting further that i can do on e1/t1 card.
Vittorio (vittogio at inwind dot it) 16 October 2005 20:45:51 when you connect the Digium PRI board to a legacy PBX via E1 trunk, if the LBO is not set at the correct value you get a RAI error (Remote Alarm Indicator) on the legacy system board.Then you need to check you exisiting PBX system for the correct value. They must be of the same range.