Likes: 0
13/05/08 15:41
Hallo,
Heb enkele weken geleden twee SIP telefoons geconfigged op onze Asterisk/FreePBX installatie.
Werkt (redelijk) perfect, maar beide toestellen gebruiken het hoofdnummer van de ISDN lijn als CID ipv. het CID nummer dat ik ingeef in FreePBX op hun extensie.
Bij de andere toestellen neemt hij wel de CID over die ik heb ingesteld per extension.
iemand?
14/05/08 08:26
Kun je de sip.conf van de lijnen laten zien?Oorspronkelijk geplaatst door TomB
14/05/08 08:51
[60]
type=friend
secret=xxxx
record_out=Never
record_in=Never
qualify=yes
port=5060
pickupgroup=1
nat=yes
mailbox=60@default
host=dynamic
dtmfmode=rfc2833
disallow=
dial=SIP/60
context=from-internal
canreinvite=no
callgroup=1
callerid=device <60>
allow=
accountcode=
[61]
type=friend
secret=xxxx
record_out=Never
record_in=Never
qualify=yes
port=5060
pickupgroup=1
nat=yes
mailbox=61@device
host=dynamic
dtmfmode=rfc2833
disallow=
dial=SIP/61
context=from-internal
canreinvite=no
callgroup=1
callerid=device <61>
allow=
accountcode=
14/05/08 09:07
Kun je je extensions.conf ook laten zien?
De SIP configuratie ziet er goed uit.
Even dubbelchecken: De bedoeling is dat interne CID gaat werken? Want het is logisch dat je als CID geen 60 of 61 de ISDN lijn op kan sturen.
14/05/08 09:51
Nee, het telefoonnummer waar het over gaat, wordt gedeeld door twee personen.
Ik geef als CID dus het volledige <nummer> in het CID veld in.
Maar Asterisk weigert die CID mee te sturen ... ze is echter exact hetzelfde opgesteld als bij de andere toestellen.
Enige verschil dat ik zie is dat alle andere toestellen gebruik maken van de nummerreeks op de 2 bestaande ISDN lijnen. En dat de toestellen (waar het niet bij werkt) het telefoonnummer van de nieuwe ISDN lijn moeten meesturen.
Heb zonet echter gemerkt dat die nieuwe ISDN lijn (die onlangs geinstalleerd is), blijkbaar niet actief is want ik krijg maar 4 channels te zien (waarschijnlijk die van de 2 reeds bestaande ISDN lijnen).
Prolly is er dus nog iets anders dat moet gewijzigd worden om die 3de poort van die DIGIUM kaart aan te spreken?
extensions.conf is http://www.briers.org/extensions.conf te vinden
Laatst gewijzigd door TomB; 14/05/08 om 09:57.
14/05/08 09:59
Zet in /etc/asterisk/logger.conf eens het volgende:
Doe dan:console => notice,warning,error,debug,verbose
En probeer daarna eens uit te bellen. Kun je van de logregels die met dat gesprek te maken hebben een Copy/Paste naar een .txt doen?asterisk -rvvvvvv
14/05/08 10:36
certain conditions. Type 'show license' for details.
================================================== =======================
Connected to Asterisk 1.2.24 currently running on asthead (pid = 4399)
Verbosity is at least 6
-- Executing Macro("SIP/60-09dbb5b0", "dialout-trunk|2|0478781981||") in new stack
-- Executing Set("SIP/60-09dbb5b0", "DIAL_TRUNK=2") in new stack
-- Executing Set("SIP/60-09dbb5b0", "DIAL_NUMBER=0478781981") in new stack
-- Executing Set("SIP/60-09dbb5b0", "ROUTE_PASSWD=") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing GotoIf("SIP/60-09dbb5b0", "0?disabletrunk|1") in new stack
-- Executing Set("SIP/60-09dbb5b0", "_NODEST=") in new stack
-- Executing Set("SIP/60-09dbb5b0", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing Set("SIP/60-09dbb5b0", "GROUP()=OUT_2") in new stack
-- Executing Macro("SIP/60-09dbb5b0", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/60-09dbb5b0", "user-callerid: device 60") in new stack
-- Executing Set("SIP/60-09dbb5b0", "AMPUSER=60") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "0?report") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "0?start") in new stack
-- Executing Set("SIP/60-09dbb5b0", "REALCALLERIDNUM=60") in new stack
-- Executing NoOp("SIP/60-09dbb5b0", "REALCALLERIDNUM is 60") in new stack
-- Executing Set("SIP/60-09dbb5b0", "AMPUSER=60") in new stack
-- Executing Set("SIP/60-09dbb5b0", "AMPUSERCIDNAME=Suzy") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "0?report") in new stack
-- Executing Set("SIP/60-09dbb5b0", "AMPUSERCID=60") in new stack
-- Executing Set("SIP/60-09dbb5b0", "CALLERID(all)="Suzy" <60>") in new stack
-- Executing Set("SIP/60-09dbb5b0", "REALCALLERIDNUM=60") in new stack
-- Executing NoOp("SIP/60-09dbb5b0", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing NoOp("SIP/60-09dbb5b0", "Using CallerID "Suzy" <60>") in new stack
-- Executing Macro("SIP/60-09dbb5b0", "record-enable|60|OUT") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/60-09dbb5b0", "recordingcheck|20080514-103422|1210754062.109") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080514-103422|1210754062.109: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/60-09dbb5b0", "No recording needed") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "0?skipoutcid") in new stack
-- Executing Set("SIP/60-09dbb5b0", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/60-09dbb5b0", "outbound-callerid|2") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/60-09dbb5b0", "REALCALLERIDNUM is 60") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/60-09dbb5b0", "USEROUTCID=<027205200>") in new stack
-- Executing Set("SIP/60-09dbb5b0", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/60-09dbb5b0", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/60-09dbb5b0", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/60-09dbb5b0", "0?report") in new stack
-- Executing Set("SIP/60-09dbb5b0", "CALLERID(all)=<027205200>") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "1?report:hidecid") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/60-09dbb5b0", "CallerID set to "" <027205200>") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing AGI("SIP/60-09dbb5b0", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/60-09dbb5b0", "OUTNUM=0478781981") in new stack
-- Executing Set("SIP/60-09dbb5b0", "custom=AMP") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,22)
-- Executing Macro("SIP/60-09dbb5b0", "dialout-trunk-predial-hook") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "0?bypass|1") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "1?customtrunk") in new stack
-- Goto (macro-dialout-trunk,s,27)
-- Executing Set("SIP/60-09dbb5b0", "pre_num=AMP:mISDN/g:first_extern/") in new stack
-- Executing Set("SIP/60-09dbb5b0", "the_num=OUTNUM") in new stack
-- Executing Set("SIP/60-09dbb5b0", "post_num=/:s:e128") in new stack
-- Executing GotoIf("SIP/60-09dbb5b0", "1?outnum:skipoutnum") in new stack
-- Goto (macro-dialout-trunk,s,31)
-- Executing Set("SIP/60-09dbb5b0", "the_num=0478781981") in new stack
-- Executing Dial("SIP/60-09dbb5b0", "mISDN/g:first_extern/0478781981/:s:e128|300|") in new stack
-- Called g:first_extern/0478781981/:s:e128
-- mISDN/1-u21 is proceeding passing it to SIP/60-09dbb5b0
-- mISDN/1-u21 is ringing
-- mISDN/1-u21 answered SIP/60-09dbb5b0
de correcte nummer zie ik wel verschijnen: <027205200>, maar deze wordt niet getoond bij hun correspondenten.
Correspondenten krijgen nummer <022630420>, wat het hoofdnummer is van de reeds bestaande ISDN lijnen.
14/05/08 10:54
Dat kan wel kloppen, want blijkbaar wordt het andere nummer niet gebruikt.
Kun je je /etc/zaptel.conf en /etc/asterisk/zapata.conf eens laten zien?
14/05/08 11:08
Een zapata.conf heb ik niet staan, enkel een zapata_conf.template
zaptel.conf
#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=<span num>,<timing source>,<line build out (LBO)>,<framing>,<coding>[,yellow]
#
# All T1/E1 spans generate a clock signal on their transmit side. The
# <timing source> parameter determines whether the clock signal from the far
# end of the T1/E1 is used as the master source of clock timing. If it is, our
# own clock will synchronise to it. T1/E1's connected directly or indirectly to
# a PSTN provider (telco) should generally be the first choice to sync to. The
# PSTN will never be a slave to you. You must be a slave to it.
#
# Choose 1 to make the equipment at the far end of the E1/T1 link the preferred
# source of the master clock. Choose 2 to make it the second choice for the master
# clock, if the first choice port fails (the far end dies, a cable breaks, or
# whatever). Choose 3 to make a port the third choice, and so on. If you have, say,
# 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each
# port should be different.
#
# 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. 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.
#
# The line build-out (or LBO) is an integer, from the following table:
# 0: 0 db (CSU) / 0-133 feet (DSX-1)
# 1: 133-266 feet (DSX-1)
# 2: 266-399 feet (DSX-1)
# 3: 399-533 feet (DSX-1)
# 4: 533-655 feet (DSX-1)
# 5: -7.5db (CSU)
# 6: -15db (CSU)
# 7: -22.5db (CSU)
#
# The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1
#
# Note: "d4" could be referred to as "sf" or "superframe"
#
# The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1
#
# E1's may have the additional keyword "crc4" to enable CRC4 checking
#
# If the keyword "yellow" follows, yellow alarm is transmitted when no
# channels are open.
#
#span=1,0,0,esf,b8zs
#span=2,1,0,esf,b8zs
#span=3,0,0,ccs,hdb3,crc4
#
# Next come the dynamic span definitions, in the form:
# dynamic=<driver>,<address>,<numchans>,<timing>
#
# Where <driver> is the name of the driver (e.g. eth), <address> is the
# driver specific address (like a MAC for eth), <numchans> is the number
# of channels, and <timing> is a timing priority, like for a normal span.
# use "0" to not use this as a timing source, or prioritize them as
# primary, secondard, etc. Note that you MUST have a REAL zaptel device
# if you are not using external timing.
#
# dynamic=eth,eth0/00:02:b3:35:43:9c,24,0
#
# Next come the definitions for using the channels. The format is:
# <device>=<channel list>
#
# Valid devices are:
#
# "e&m" : Channel(s) are signalled using E&M signalling (specific
# implementation, such as Immediate, Wink, or Feature Group D
# are handled by the userspace library).
# "fxsls" : Channel(s) are signalled using FXS Loopstart protocol.
# "fxsgs" : Channel(s) are signalled using FXS Groundstart protocol.
# "fxsks" : Channel(s) are signalled using FXS Koolstart protocol.
# "fxols" : Channel(s) are signalled using FXO Loopstart protocol.
# "fxogs" : Channel(s) are signalled using FXO Groundstart protocol.
# "fxoks" : Channel(s) are signalled using FXO Koolstart protocol.
# "sf" : Channel(s) are signalled using in-band single freq tone.
# Syntax as follows:
# channel# => sf:<rxfreq>,<rxbw>,<rxflag>,<txfreq>,<txlevel>,<tx flag>
# rxfreq is rx tone freq in hz, rxbw is rx notch (and decode)
# bandwith in hz (typically 10.0), rxflag is either 'normal' or
# 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone
# level in dbm, txflag is either 'normal' or 'inverted'. Set
# rxfreq or txfreq to 0.0 if that tone is not desired.
# "unused" : No signalling is performed, each channel in the list remains idle
# "clear" : Channel(s) are bundled into a single span. No conversion or
# signalling is performed, and raw data is available on the master.
# "indclear": Like "clear" except all channels are treated individually and
# are not bundled. "bchan" is an alias for this.
# "rawhdlc" : The zaptel driver performs HDLC encoding and decoding on the
# bundle, and the resulting data is communicated via the master
# device.
# "fcshdlc" : The zapdel driver performs HDLC encoding and decoding on the
# bundle and also performs incoming and outgoing FCS insertion
# and verification. "dchan" is an alias for this.
# "nethdlc" : The zaptel driver bundles the channels together into an
# hdlc network device, which in turn can be configured with
# sethdlc (available separately).
# "dacs" : The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon
# "dacsrbs" : The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon and
# also performs the DACSing of RBS bits
#
# The channel list is a comma-separated list of channels or ranges, for
# example:
#
# 1,3,5 (channels one, three, and five)
# 16-23, 29 (channels 16 through 23, as well as channel 29
#
# So, some complete examples are:
# e&m=1-12
# nethdlc=13-24
# fxsls=25,26,27,28
# fxols=29-32
#
#fxoks=1-24
#bchan=25-47
#dchan=48
#fxols=1-12
#fxols=13-24
#e&m=25-29
#nethdlc=30-33
#clear=44
#clear=45
#clear=46
#clear=47
#fcshdlc=48
#dacs=1-24:48
#dacsrbs=1-24:48
#
# Finally, you can preload some tone zones, to prevent them from getting
# overwritten by other users (if you allow non-root users to open /dev/zap/*
# interfaces anyway. Also this means they won't have to be loaded at runtime.
# The format is "loadzone=<zone>" where the zone is a two letter country code.
#
# You may also specify a default zone with "defaultzone=<zone>" where zone
# is a two letter country code.
#
# An up-to-date list of the zones can be found in the file zaptel/zonedata.c
#
loadzone = us
#loadzone = us-old
#loadzone=gr
#loadzone=it
#loadzone=fr
#loadzone=de
#loadzone=uk
#loadzone=fi
#loadzone=jp
#loadzone=sp
#loadzone=no
#loadzone=hu
#loadzone=lt
#loadzone=pl
defaultzone=us
#
# Section for PCI Radio Interface
# (see http://www.zapatatelephony.org/app_rpt.html)
#
# The PCI Radio Interface card interfaces up to 4 two-way radios (either
# a base/mobile radio or repeater system) to Zaptel channels. The driver
# may work either independent of an application, or with it, through
# the driver;s ioctl() interface. This file gives you access to specify
# load-time parameters for Radio channels, so that the driver may run
# by itself, and just act like a generic Zaptel radio interface.
#
# Unlike the rest of this file, you specify a block of parameters, and
# then the channel(s) to which they apply. CTCSS is specified as a frequency
# in tenths of hertz, for example 131.8 HZ is specified as 1318. DCS
# for receive is specified as the code directly, for example 223. DCS for
# transmit is specified as D and then the code, for example D223.
#
# The hardware supports a "community" CTCSS decoder system that has
# arbitrary transmit CTCSS or DCS codes associated with them, unlike
# traditional "community" systems that encode the same tone they decode.
#
# this example is a single tone DCS transmit and receive
#
# # specify the transmit tone (in DCS mode this stays constant)
# tx=D371
# # specify the receive DCS code
# dcsrx=223
#
# this example is a "community" CTCSS (if you only want a single tone, then
# only specify 1 in the ctcss list)
#
# # specify the default transmit tone (when not receiving)
# tx=1000
# # Specify the receive freq, the tag (use 0 if none), and the transmit code.
# # The tag may be used by applications to determine classification of tones.
# # The tones are to be specified in order of presedence, most important first.
# # Currently, 15 tones may be specified..
# ctcss=1318,1,1318
# ctcss=1862,1,1862
#
# The following parameters may be omitted if their default value is acceptible
#
# # set the receive debounce time in milliseconds
# debouncetime=123
# # set the transmit quiet dropoff burst time in milliseconds
# bursttime=234
# # set the COR level threshold (specified in tenths of millivolts)
# # valid values are {3125,6250,9375,12500,15625,18750,21875,25000}
# corthresh=12500
# # Invert COR signal {y,n}
# invertcor=y
# # set the external tone mode; yes, no, internal {y,n,i}
# exttone=y
#
# Now apply the configuration to the specified channels:
#
# # We are all done with our channel parameters, so now we specify what
# # channels they apply to
# channels=1-4
Is het niet de MISDN.conf dat je nodig hebt?
[root@asthead etc]# cat misdn-init.conf
#
# Configuration file for your misdn hardware
#
# Usage: /usr/sbin/misdn-init start|stop|restart|config|scan|help
#
#
# Card Settings
#
# Syntax: card=<number>,<type>[,<option>...]
#
# <number> count your cards beginning with 1
# <type> either 0x1,0x4 or 0x8 for your hfcmulti hardware,
# or the name of your card driver module.
# <option> ulaw - uLaw (instead of aLaw)
# dtmf - enable DTMF detection on all B-channels
#
# pcm_slave - set PCM bus into slave mode
# If you have a set of cards, all wired via PCM. Set
# all cards into pcm_slave mode and leave one out.
# The left card will automatically be Master.
#
# ignore_pcm_frameclock - this can be set in conjunction with
# pcm_slave. If this card has a
# PCI Bus Position before the Position
# of the Master, then this port cannot
# yet receive a frameclock, so it must
# ignore the pcm frameclock.
#
# rxclock - use clocking for pcm from ST Port
# crystalclock - use clocking for pcm from PLL (genrated on board)
# watchdog - This dual E1 Board has a Watchdog for
# transparent mode
#
#
card=1,0x4
#
# Port settings
#
# Syntax: <port_type>=<port_number>[,<port_number>...]
#
# <port_type> te_ptp - TE-Mode, PTP
# te_ptmp - TE-Mode, PTMP
# te_capi_ptp - TE-Mode (capi), PTP
# te_capi_ptmp - TE-Mode (capi), PTMP
# nt_ptp - NT-Mode, PTP
# nt_ptmp - NT-Mode, PTMP
# <port_number> port that should be considered
#
te_ptp=1,2,3
#
# Port Options
#
# Syntax: option=<port_number>,<option>[,<option>...]
#
# <option> master_clock - use master clock for this S/T interface
# (only once per chip, only for HFC 8/4)
# optical - optical (only HFC-E1)
# los - report LOS (only HFC-E1)
# ais - report AIS (only HFC-E1)
# slip - report SLIP (only HFC-E1)
# nocrc4 - turn off crc4 mode use double frame instead
# (only HFC-E1)
#
# The master_clock option is essential for retrieving and transmitting
# faxes to avoid failures during transmission. It tells the driver to
# synchronize the Card with the given Port which should be a TE Port and
# connected to the PSTN in general.
#
#option=1,master_clock
#option=2,ais,nocrc4
#option=3,optical,los,ais,slip
#
# General Options for your hfcmulti hardware
#
# poll=<number>
#
# Only one poll value must be given for all cards.
# Give the number of samples for each fifo process.
# By default 128 is used. Decrease to reduce delay, increase to
# reduce cpu load. If unsure, don't mess with it!!!
# Valid is 32, 64, 128, 256.
#
# dsp_poll=<number>
# This is the poll option which is used by mISDN_dsp, this might
# differ from the one given by poll= for the hfc based cards, since
# they can only use multiples of 32, the dsp_poll is dependant on
# the kernel timer setting which can be found in the CPU section
# in the kernel config. Defaults are there either 100Hz, 250Hz
# or 1000Hz. If your setting is either 1000 or 250 it is compatible
# with the poll option for the hfc chips, if you have 100 it is
# different and you need here a multiple of 80.
# The default is to have no dsp_poll option, then the dsp itself
# finds out which option is the best to use by itself
#
# pcm=<number>
#
# Give the id of the PCM bus. All PCM busses with the same ID
# are expected to be connected and have equal slots.
# Only one chip of the PCM bus must be master, the others slave.
#
# debug=<number>
#
# Enable debugging (see hfc_multi.h for debug options).
#
# dsp_options=<number>
#
# set this to 2 and you'll have software bridging instead of
# hardware bridging.
#
#
# dtmfthreshold=<milliseconds>
#
# Here you can tune the sensitivity of the dtmf tone recognizer.
#
# timer=<1|0>
#
# set this to 1 if you want hfcmulti to register at ztdummy (zaptel)
# and provide a 1khz timing source for it. This makes it possible
# to have an accurate timing source for asterisk through zaptel from
# hfcmulti to make applications like Meetme and faxing between wctdm
# and hfcmulti work properly.
#
poll=128
#dsp_poll=128
#dsp_options=0
#dtmfthreshold=100
debug=0
We hebben een probleem dat een call soms geen sound doorgeeft (zowel ontvanger als beller). Dit komt zowel intern als extern voor in 5% à 10% van alle calls.
Volgende items heb ik deze morgen net gewijzigd, om te proberen dat prob op te lossen:
--> dsp_poll, dsp_options en dtmfthreshold net in comment gezet.
--> Bij "card=1,0x4", stond er eerst "card=1,0x4, rxclock", wat ik deze morgen ook gewijzigd heb.
Laatst gewijzigd door TomB; 14/05/08 om 11:17.
14/05/08 11:14
Wat voor kaart heb je in je server zitten?
14/05/08 12:13
Digium B410P - 4 poort ISDN
ben trouwens nog aan het zoeken wat ik achter die "card=1,0x4" moet zetten, want daar zijn ettelijke opties voor.
stond op rxclock, maar is dat wel de correcte?
14/05/08 12:54
Heb je een /etc/misdn-init.conf ?
Zo ja, kunnen we daar naar kijken?
14/05/08 13:46
yep, dacht dat je die bedoelde, dus had die reeds gepost hierboven.
Hier nog eens:
[root@asthead etc]# cat misdn-init.conf
#
# Configuration file for your misdn hardware
#
# Usage: /usr/sbin/misdn-init start|stop|restart|config|scan|help
#
#
# Card Settings
#
# Syntax: card=<number>,<type>[,<option>...]
#
# <number> count your cards beginning with 1
# <type> either 0x1,0x4 or 0x8 for your hfcmulti hardware,
# or the name of your card driver module.
# <option> ulaw - uLaw (instead of aLaw)
# dtmf - enable DTMF detection on all B-channels
#
# pcm_slave - set PCM bus into slave mode
# If you have a set of cards, all wired via PCM. Set
# all cards into pcm_slave mode and leave one out.
# The left card will automatically be Master.
#
# ignore_pcm_frameclock - this can be set in conjunction with
# pcm_slave. If this card has a
# PCI Bus Position before the Position
# of the Master, then this port cannot
# yet receive a frameclock, so it must
# ignore the pcm frameclock.
#
# rxclock - use clocking for pcm from ST Port
# crystalclock - use clocking for pcm from PLL (genrated on board)
# watchdog - This dual E1 Board has a Watchdog for
# transparent mode
#
#
card=1,0x4
#
# Port settings
#
# Syntax: <port_type>=<port_number>[,<port_number>...]
#
# <port_type> te_ptp - TE-Mode, PTP
# te_ptmp - TE-Mode, PTMP
# te_capi_ptp - TE-Mode (capi), PTP
# te_capi_ptmp - TE-Mode (capi), PTMP
# nt_ptp - NT-Mode, PTP
# nt_ptmp - NT-Mode, PTMP
# <port_number> port that should be considered
#
te_ptp=1,2,3
#
# Port Options
#
# Syntax: option=<port_number>,<option>[,<option>...]
#
# <option> master_clock - use master clock for this S/T interface
# (only once per chip, only for HFC 8/4)
# optical - optical (only HFC-E1)
# los - report LOS (only HFC-E1)
# ais - report AIS (only HFC-E1)
# slip - report SLIP (only HFC-E1)
# nocrc4 - turn off crc4 mode use double frame instead
# (only HFC-E1)
#
# The master_clock option is essential for retrieving and transmitting
# faxes to avoid failures during transmission. It tells the driver to
# synchronize the Card with the given Port which should be a TE Port and
# connected to the PSTN in general.
#
#option=1,master_clock
#option=2,ais,nocrc4
#option=3,optical,los,ais,slip
#
# General Options for your hfcmulti hardware
#
# poll=<number>
#
# Only one poll value must be given for all cards.
# Give the number of samples for each fifo process.
# By default 128 is used. Decrease to reduce delay, increase to
# reduce cpu load. If unsure, don't mess with it!!!
# Valid is 32, 64, 128, 256.
#
# dsp_poll=<number>
# This is the poll option which is used by mISDN_dsp, this might
# differ from the one given by poll= for the hfc based cards, since
# they can only use multiples of 32, the dsp_poll is dependant on
# the kernel timer setting which can be found in the CPU section
# in the kernel config. Defaults are there either 100Hz, 250Hz
# or 1000Hz. If your setting is either 1000 or 250 it is compatible
# with the poll option for the hfc chips, if you have 100 it is
# different and you need here a multiple of 80.
# The default is to have no dsp_poll option, then the dsp itself
# finds out which option is the best to use by itself
#
# pcm=<number>
#
# Give the id of the PCM bus. All PCM busses with the same ID
# are expected to be connected and have equal slots.
# Only one chip of the PCM bus must be master, the others slave.
#
# debug=<number>
#
# Enable debugging (see hfc_multi.h for debug options).
#
# dsp_options=<number>
#
# set this to 2 and you'll have software bridging instead of
# hardware bridging.
#
#
# dtmfthreshold=<milliseconds>
#
# Here you can tune the sensitivity of the dtmf tone recognizer.
#
# timer=<1|0>
#
# set this to 1 if you want hfcmulti to register at ztdummy (zaptel)
# and provide a 1khz timing source for it. This makes it possible
# to have an accurate timing source for asterisk through zaptel from
# hfcmulti to make applications like Meetme and faxing between wctdm
# and hfcmulti work properly.
#
poll=128
#dsp_poll=128
#dsp_options=0
#dtmfthreshold=100
debug=0
14/05/08 15:01
Kun je eigenlijk vanaf een toestel dat nieuwe nummer bellen? Zo ja: komt deze ook binnen op je Asterisk?
15/05/08 15:40
yep, binnenbellen werkt zonder problemen ... dus is wel degelijk correct geinstalleerd, verschijnt enkel niet in men Asterisk/freepbx als trunk.
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