DAHDI +Freepbx13

good evening
I can not solve a simple problem.

I have:
freepbx13 (latest)

RPI CARD (E1) / 2 ports
http://parabel-labs.com/products/quasar-mini

yum -y install kernel-devel-$(uname -r) libtool* make gcc patch perl bison gcc-c++ ncurses-devel flex libtermcap-devel autoconf* automake* autoconf
wget http://parabel.ru/d/software/dahdi/dahdi_2.11.1%2B2.11.1-parabel_2.11.1.7.tar.bz2
tar xfv dahdi_2.11.1%2B2.11.1-parabel_2.11.1.7.tar.bz2
cd dahdi_2.11.1%2B2.11.1-parabel_2.11.1.7.tar.bz2
./build.sh
./install.sh

dahdi show status
Description Alarms IRQ bpviol CRC Fra Codi Options LBO
Parabel Quasar 0 Span 1 OK 0 0 0 CCS HDB3 0 db (CSU)/0-133 feet (DSX-1)
Parabel Quasar 0 Span 2 RED 0 1 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1)

i`am installing RPI card.
pri show spans
PRI span 1/0: Up, Active (one link up) (link to Provider 1E switch)

dahdi show channels
Chan Extension Context Language MOH Interpret Blocked In Service Description _
_ pseudo default default Yes _
_ 1 from-digital ru default Yes _
_ 2 from-digital ru default Yes _
_ 3 from-digital ru default Yes _
_ 4 from-digital ru default Yes

…5-30
_ 31 from-digital ru default Yes_

From this I draw conclusions that the card is fully operational. Is seen as an asterisk.
(Before that I installed it, and the driver)

QUESTION:

  1. HOW DO I BUILD THE TRUNK ON FreePBX GUI!?

1.The manufacturer gave configuration files for Asterisk (non Freepbx)
- /etc/dahdi/modules:
quasar
quasarm

- /etc/dahdi/system.conf
# If you choose 0, the port will never be used as a source of timing. This is
# appropriate when you know the far end should always be a slave to you. If
# the port is connected to a channel bank, for example, you should always be
# its master. Likewise, BRI TE ports should always be configured as a slave.
# Any number of ports can be marked as 0.
#
# Incorrect timing sync may cause clicks/noise in the audio, poor quality or failed
# faxes, unreliable modem operation, and is a general all round bad thing.

# SPAN 1, timing source (E1 slave), HDB3, CRC4, CCS.
span=1,1,0,ccs,hdb3,crc4

# SPAN 2, timing target (E1 master), HDB3, CRC4, CCS.
span=2,0,0,ccs,hdb3,crc4

# SPAN 1, CCS(PRI/SS7/…), A-law, OSLEC echocanceller.
bchan=1-15,17-31
dchan=16
alaw=1-15,17-31
echocanceller=oslec,1-15,17-31

# SPAN 2, CCS(PRI/SS7/…), A-law, OSLEC echocanceller.
bchan=32-46,48-62
dchan=47
alaw=32-46,48-62
echocanceller=oslec,32-46,48-62

# Setting correct zone info (tone info)
loadzone=ru
defaultzone=ru

- EXtension file - i think its o outgoin
[general]
static=yes
writeprotect=yes
autofallthrough=yes
clearglobalvars=no
;priorityjumping=no

[globals]
CONSOLE=Console/dsp

[default]
exten => s,1,Macro(mylog, “BUG: Unhandled context !!!”)
exten => s,n,Hangup()
_exten => X.,1,Macro(mylog, “BUG: Unhandled context !!!”)
_exten => X.,n,Hangup()

[test_context]
_exten => 1.,1,Wait(0)
; PSTN can reject an outgoing call without caller id.
_;exten => 1.,n,Set(CALLERID(all)=84950000000)
_exten => 1.,n, Dial(DAHDI/g0/${EXTEN:1})
_exten => 1.,n,Hangup()

_exten => 2.,1,Wait(0)
; Our PBX should send caller ID to the client PBX
_;exten => 2.,n,Set(CALLERID(all)=84950000001)
_exten => 2.,n, Dial(DAHDI/g1/${EXTEN:1})
_exten => 2.,n,Hangup()

exten => 310,1, Dial(SIP/sip1)
exten => 320,1, Dial(SIP/sip2)

exten => 91,1, Answer
exten => 91,n, Wait(3600)

exten => 92,1, Answer
exten => 92,n, Echo()

exten => 93,1,Answer
exten => 93,n,MP3Player(/etc/asterisk/1.mp3)

[pri_base]
_exten => X.,1,Answer()
_exten => X.,n,Verbose(Received callerID: ${CALLERID(all)})
_exten => X.,n,Echo()
_;exten => X.,n,MP3Player(/etc/asterisk/1.mp3)
_exten => X.,n,Hangup()

exten => s,1,Answer()
exten => s,n,Verbose(Received callerID: ${CALLERID(all)})
exten => s,n,Echo()
;exten => s,n,MP3Player(/etc/asterisk/1.mp3)
exten => s,n,Hangup()

[pri0]
include => pri_base

[pri1]
include => pri_base

  • and CHAIN_DAHDI.conf
    [channels]
    relaxdtmf=yes ; pavel - necessary on bad lines…
    rxflash=850

;== busy
busydetect=yes
busycount=3

;== pulse
pulsedial=no
pulse=no

;== Calls handling ==
callwaiting=yes ;If enabled, Asterisk will generate “call waiting pips” when you are already in a conversation
;{
threewaycalling=yes
transfer=yes
;}
cancallforward=yes
immediate=no ;yes to let Asterisk handle ‘s’ extention (passing control immediately to Asterisk)

;== Echo ==
echocancel = yes
echocancelwhenbridged = yes
echotraining=no

;== Caller ID ==
usecallerid=yes
callerid=asreceived
;Unrecognized prilocaldialplan NPI modifier: s
;don’t have a time to find the reason. disabling…
hidecallerid=yes
;;callwaitingcallerid=yes
useincomingcalleridonzaptransfer=yes
;
cidsignalling=bell
cidstart=ring
restrictcid=no
callreturn=yes ;dial *69 to have Asterisk read to you the caller ID of the last person to call
pridialplan=national
prilocaldialplan=national
internationalprefix=
nationalprefix=
_localprefix= _

;==== FXO lines (external PBX) ====
group=0
context=pri0
signalling=pri_cpe
switchtype=euroisdn
channel => 1-15,17-31
_group = _
context = default

group=1
context=pri1
signalling=pri_net
switchtype=euroisdn
channel => 32-46,48-62
_group = _
context = default

  1. HOW DO I BUILD THE TRUNK ON FreePBX GUI!?

There are too many errors in what you have posted to expect your system to work under FreeBX, there would NEVER be any"_" character in any of the files you posted , unless you know what you are doing, CHAIN_DAHDI.conf just won’t work . . . . You will also have to explain what you mean by “EXtension file”

The dahdi helper module is very limited in what it supports, almost certainly not your hardware though .

i`ts no error config… its “bug” this forum =))

_group = _
this symbol " _ " - text format - Emphasis = Emphasis

About the configuration. I want to understand how to me the current text settings, FIND in the WEB GUI freepbx and expose them.

try posting it all again surrounding it with triple back-ticks and also check your spelling and case of your letters, for example if your CHAIN_DAHDI.conf should really be chan_dahdi.conf then it is unlikely that you have a valid

context=default

You probably should be using

from-digital
from-pstn
or
from-trunk
(effectively they are all the same)

or perhaps

from-did-direct

which most E1/T1’s deployments find more efficient

All are documented in /etc/asterisk/extensions.conf