Asterisk Bristugg 1.2.19 mit mISDN Treiber Interner S0 funktioniert nicht

[sebastianoh]

Hallo Gemeinde,

ich habe folgendes Problem:

Mein Asterisk Bristuff 1.2.19 mit Beronet 4 S0 Karte läuft soweit einwandfrei nur der interne S0 an Port 4 lässt keine Anrufe raus. Eingehende Anrufe werden an den internen S0 geroutet und signalisiert.

Habe hier mal die Fehlermeldung bei der Anwahl über den internen S0 angefügt und ebenfalls die weiteren erforderlichen Dateien.

Ausgabe über Asterisk -vvvvvvc

*CLI> -- Executing Dial("mISDN/4-1", "misdn/g:Anlagenanschluss/017") in new stack
-- Called g:Anlagenanschluss/017
Dec 11 16:54:46 WARNING[27736]: pbx.c:2271 __ast_pbx_run: mISDN/4-1 already has PBX structure??
-- Executing Dial("mISDN/4-1", "misdn/g:Anlagenanschluss/0172") in new stack
-- Called g:Anlagenanschluss/0172
Dec 11 16:54:46 WARNING[27736]: channel.c:1653 ast_waitfor_nandfds: Thread -1241187440 Blocking 'mISDN/4-1', already blocked by thread -1240921200 in procedure ast_waitfor_nandfds
Dec 11 16:54:46 WARNING[27742]: pbx.c:2271 __ast_pbx_run: mISDN/4-1 already has PBX structure??
-- Executing Dial("mISDN/4-1", "misdn/g:Anlagenanschluss/01724") in new stack
-- Called g:Anlagenanschluss/01724
Dec 11 16:54:46 WARNING[27742]: channel.c:1653 ast_waitfor_nandfds: Thread -1241453680 Blocking 'mISDN/4-1', already blocked by thread -1240921200 in procedure ast_waitfor_nandfds
Dec 11 16:54:46 WARNING[27748]: pbx.c:2271 __ast_pbx_run: mISDN/4-1 already has PBX structure??
-- Executing Dial("mISDN/4-1", "misdn/g:Anlagenanschluss/017245") in new stack
-- Called g:Anlagenanschluss/017245
Dec 11 16:54:46 WARNING[27748]: channel.c:1653 ast_waitfor_nandfds: Thread -1241719920 Blocking 'mISDN/4-1', already blocked by thread -1241187440 in procedure ast_waitfor_nandfds
Dec 11 16:54:46 WARNING[27754]: pbx.c:2271 __ast_pbx_run: mISDN/4-1 already has PBX structure??
-- Executing Dial("mISDN/4-1", "misdn/g:Anlagenanschluss/0172454") in new stack
-- Called g:Anlagenanschluss/0172454
Dec 11 16:54:46 WARNING[27754]: channel.c:1653 ast_waitfor_nandfds: Thread -1241986160 Blocking 'mISDN/4-1', already blocked by thread -1241187440 in procedure ast_waitfor_nandfds
Dec 11 16:54:46 WARNING[27760]: pbx.c:2271 __ast_pbx_run: mISDN/4-1 already has PBX structure??
-- Executing Dial("mISDN/4-1", "misdn/g:Anlagenanschluss/01724544") in new stack
-- Called g:Anlagenanschluss/01724544
Dec 11 16:54:46 WARNING[27760]: channel.c:1653 ast_waitfor_nandfds: Thread -1242252400 Blocking 'mISDN/4-1', already blocked by thread -1241187440 in procedure ast_waitfor_nandfds
Dec 11 16:54:46 WARNING[27766]: pbx.c:2271 __ast_pbx_run: mISDN/4-1 already has PBX structure??
-- Executing Dial("mISDN/4-1", "misdn/g:Anlagenanschluss/017245447") in new stack
Dec 11 16:54:46 WARNING[27766]: chan_misdn.c:2897 misdn_request: Could not Dial out on group 'Anlagenanschluss'.
Either the L2 and L1 on all of these ports where DOWN (see 'show application misdn_check_l2l1')
Or there was no free channel on none of the ports

Dec 11 16:54:46 NOTICE[27766]: app_dial.c:1096 dial_exec_full: Unable to create channel of type 'misdn' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
Dec 11 16:54:46 WARNING[27766]: channel.c:1653 ast_waitfor_nandfds: Thread -1242518640 Blocking 'mISDN/4-1', already blocked by thread -1241187440 in procedure ast_waitfor_nandfds
Dec 11 16:54:47 WARNING[27770]: pbx.c:2271 __ast_pbx_run: mISDN/4-1 already has PBX structure??
== Starting mISDN/4-1 at hfc-incoming,0172454473,2 failed so falling back to exten 's'
== Starting mISDN/4-1 at hfc-incoming,s,2 still failed so falling back to context 'default'
Dec 11 16:54:47 WARNING[27770]: pbx.c:2378 __ast_pbx_run: Channel 'mISDN/4-1' sent into invalid extension 's' in context 'default', but no invalid handler
Dec 11 16:54:47 WARNING[27770]: channel.c:1409 ast_hangup: Hard hangup called by thread -1242784880 on mISDN/4-1, while fd is blocked by thread -1241187440 in procedure ast_waitfor_nandfds! Expect a failure
== Spawn extension (default, s, 1) exited non-zero on ''
Dec 11 16:54:47 ERROR[27736]: channel.c:942 ast_channel_free: Unable to find channel in list to free. Assuming it has already been done.
== Spawn extension (default, s, 1) exited non-zero on ''
Segmentation fault






Ausgabe über Asterisk -r

Dec 11 16:59:50 WARNING[27927]: pbx.c:2271 __ast_pbx_run: mISDN/4-1 already has PBX structure??
Dec 11 16:59:50 WARNING[27927]: pbx.c:2378 __ast_pbx_run: Channel 'mISDN/4-1' se nt into invalid extension 's' in context 'default', but no invalid handler
Dec 11 16:59:50 WARNING[27927]: channel.c:1409 ast_hangup: Hard hangup called by thread -1241244784 on mISDN/4-1, while fd is blocked by thread -1241949296 in p rocedure ast_waitfor_nandfds! Expect a failure
Dec 11 16:59:50 ERROR[27926]: channel.c:942 ast_channel_free: Unable to find cha nnel in list to free. Assuming it has already been done.
beronet*CLI>

misdn-init.conf:

misdn-init.conf
Page 1 of 2
#
# 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
nt_ptmp=4
#
# 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
misdn-init.conf
Page 2 of 2
# 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

misdn.conf:

[general]
bridging=yes
; ISDN eingehend Ports 1-3
[Anlagenanschluss]
ports=1,2,3
context=isdn-incoming
msns=*
; ISDN Mehrger�eanschluss intern
[Intern]
ports=4
context=hfc-incoming
msns=*

extensions.conf:

extensions.conf
Page 1 of 4
[general]
static=yes
writeprotect=no
priorityjumping=yes
[globals]
IAXINFO=guest
TCOM=misdn/g:Anlagenanschluss
FAX=misdn/g:Intern
[default]
; kurzwahlen etc.
[calls]
; Record voice file to /tmp directory
exten => *41,1,Wait(1) ; Call 41 to Record new Sound Files
exten => *41,2,Record(/tmp/asterisk-recording:ulaw) ; Press # to stop recording
exten => *41,3,Wait(1)
exten => *41,4,Playback(/tmp/asterisk-recording) ; Listen to your voice
exten => *41,5,wait(2)
exten => *41,6,Hangup
; test moh on 42
exten => *42,1,Answer
exten => *42,2,MusicOnHold()
; echo-test on 43
;exten => *43,1,Playback(demo-echotest)
exten => *43,1,Playback(beep)
exten => *43,2,Echo
exten => *43,3,Playback(demo-echodone)
exten => *43,4,Hangup
; zeitansage
exten => *49,1,DateTime
; local channel to wait before ring group
exten => *950,1,GotoIf($[${LEN(${CALLERIDNUM})} = 2]?501) ; es wird von einer nebenstelle
angerufen
exten => *950,2,Wait(8)
exten => *950,3,Dial(SIP/10&SIP/20&SIP/30&SIP/40&SIP/60&SIP/70,40,wWt)
exten => *950,501,Busy
; local channel to toggle light off after TFE
exten => *951,1,Wait(4)
exten => *951,2,DevState(${EXTEN},0) ; lights off!
exten => _*905XX,1,Dial(SIP/${EXTEN:4},3600,m(none)d)
exten => _*905XX,n,Hangup
; *902${EXTEN} setzt CLIP
exten => *90210,hint,DS/*90210
exten => *90220,hint,DS/*90220
exten => *90230,hint,DS/*90230
exten => *90240,hint,DS/*90240
exten => *90250,hint,DS/*90250
exten => *90260,hint,DS/*90260
exten => *90270,hint,DS/*90270
exten => _*902XX,1,Answer
exten => _*902XX,2,DBGet(temp=Funktionen/CLIP-${EXTEN:4})
exten => _*902XX,103,DBput(Funktionen/CLIP-${EXTEN:4}=1) ; clip war nicht gesetzt, jetzt
setzen
exten => _*902XX,104,DevState(${EXTEN},2) ; lights on!
exten => _*902XX,105,Wait(1)
exten => _*902XX,106,Hangup
exten => _*902XX,3,DBdel(Funktionen/CLIP-${EXTEN:4}) ; cclip war gesetzt, jetzt loeschen
exten => _*902XX,4,DevState(${EXTEN},0) ; lights off!
exten => _*902XX,5,Wait(1)
exten => _*902XX,6,Hangup
exten => t,1,Busy
; incoming ISDN BRI calls arrive in this context
; we need an extension for every DID/MSN or a pattern that will match all
[isdn-incoming]
exten => _325444XX,1,Goto(sip-phones,${EXTEN:6},1)
; eingehend, zentrale, isdn: 3254440
exten => 3254440,1,Goto(zentrale,s,1)
exten => 0,1,Goto(zentrale,s,1)
exten => 00,1,Goto(zentrale,s,1)
exten => s,1,ResponseTimeout(5)
exten => s,2,DigitTimeout(5)
;exten => s,3,Goto(zentrale,s,1)
extensions.conf
Page 2 of 4
; in sip.conf you put context=sip-phones so the phones can dial out
; using zaptel group 1 (which includes all BRI ports, see zapata.conf)
exten => i,1,Congestion
exten => h,1,NoOp
exten => t,1,Goto(zentrale,s,1)
exten => o,1,Goto(zentrale_ivr,s,1)
[hfc-incoming]
; dial from ISTEC TFE to all
exten => 75,1,Goto(zentrale,s,1)
; ISDN Dialout Fax, Data, etc.
;exten => _XX.,1,SetCallerid(325444${CALLERIDNUM})
exten => _XX.,1,Dial(misdn/g:Anlagenanschluss/${EXTEN})
exten => _XX.,2,Hangup
; if the called party is busy
exten => _XX.,103,Playtones(busy)
exten => _XX.,104,Wait(10)
exten => _XX.,105,Hangup
; if all zaptel channels in that group are in use
; or the D channels are down
exten => _XX.,203,Playtones(congestion)
exten => _XX.,204,Wait(10)
exten => _XX.,205,Hangup
exten => i,1,Congestion
exten => h,1,NoOp
exten => t,1,Busy
exten => o,1,Goto(zentrale_ivr,s,1)
[sip-phones]
include => calls
;--------------------------------------------------------------
; SIP phones nur bei interner anwahl
exten => 10,hint,SIP/10
exten => 10,1,Dial(SIP/10&LOCAL/*950@calls,50,wWt)
exten => 10,2,Hangup
exten => 10,102,GotoIf($[${LEN(${CALLERIDNUM})} = 2]?501) ;
exten => 10,103,Playtones(busy)
exten => 10,104,Wait(2)
exten => 10,105,Hangup(17)
exten => 10,501,Hangup
exten => 20,hint,SIP/20
exten => 20,1,Dial(SIP/20&LOCAL/*950@calls,50,wWt)
exten => 20,2,Hangup
exten => 20,102,GotoIf($[${LEN(${CALLERIDNUM})} = 2]?501) ;
exten => 20,103,Playtones(busy)
exten => 20,104,Wait(2)
exten => 20,105,Hangup(17)
exten => 20,501,Hangup
exten => 30,hint,SIP/30
exten => 30,1,Dial(SIP/30&LOCAL/*950@calls,50,wWt)
exten => 30,2,Hangup
exten => 30,102,GotoIf($[${LEN(${CALLERIDNUM})} = 2]?501) ;
exten => 30,103,Playtones(busy)
exten => 30,104,Wait(2)
exten => 30,105,Hangup(17)
exten => 30,501,Hangup
exten => 40,hint,SIP/40
exten => 40,1,Dial(SIP/40&LOCAL/*950@calls,50,wWt)
exten => 40,2,Hangup
exten => 40,102,GotoIf($[${LEN(${CALLERIDNUM})} = 2]?501) ;
exten => 40,103,Playtones(busy)
exten => 40,104,Wait(2)
exten => 40,105,Hangup(17)
exten => 40,501,Hangup
exten => 50,hint,SIP/50
exten => 50,1,Dial(SIP/50&LOCAL/*950@calls,50,wWt)
exten => 50,2,Hangup
exten => 50,102,GotoIf($[${LEN(${CALLERIDNUM})} = 2]?501) ;
exten => 50,103,Playtones(busy)
exten => 50,104,Wait(2)
exten => 50,105,Hangup(17)
exten => 50,501,Hangup
exten => 60,hint,SIP/60
exten => 60,1,Dial(SIP/60&LOCAL/*950@calls,50,wWt)
exten => 60,2,Hangup
extensions.conf
Page 3 of 4
exten => 60,102,GotoIf($[${LEN(${CALLERIDNUM})} = 2]?501) ;
exten => 60,103,Playtones(busy)
exten => 60,104,Wait(2)
exten => 60,105,Hangup(17)
exten => 60,501,Hangup
exten => 70,hint,SIP/70
exten => 70,1,Dial(SIP/70&LOCAL/*950@calls,50,wWt)
exten => 70,2,Hangup
exten => 70,102,GotoIf($[${LEN(${CALLERIDNUM})} = 2]?501) ;
exten => 70,103,Playtones(busy)
exten => 70,104,Wait(2)
exten => 70,105,Hangup(17)
exten => 70,501,Hangup
; Fax goes to g2
exten => 34,1,Settransfercapability(SPEECH)
exten => 34,2,Dial(misdn/g:Intern/34,30,wWt)
exten => 34,3,Hangup
exten => 34,103,Playtones(busy)
exten => 34,104,Wait(2)
exten => 34,105,Hangup(17)
; ISDN Data #1
exten => 71,1,Dial(misdn/g:Intern/71,30,wWt)
exten => 71,2,Hangup
exten => 71,102,Playtones(busy)
exten => 71,103,Wait(2)
exten => 71,104,Hangup(17)
; ISDN Data #2
exten => 72,1,Dial(misdn/g:Intern/72,30,wWt)
exten => 72,2,Hangup
exten => 72,102,Playtones(busy)
exten => 72,103,Wait(2)
exten => 72,104,Hangup(17)
; ISDN TFE
exten => 75/_XX,hint,DS/75
exten => 75/_XX,1,DevState(${EXTEN},2) ; lights on!
exten => 75/_XX,2,Dial(misdn/g:Intern/75,30,S(8)D(ww*71#1111*))
exten => 75/_XX,3,DevState(${EXTEN},0) ; lights off!
exten => 75/_XX,4,Hangup
exten => 75/_XX,103,Playtones(busy)
exten => 75/_XX,104,Wait(2)
exten => 75/_XX,105,Hangup(17)
; ISDN TFE einrichtung
exten => 76,hint,DS/76
exten => 76,1,Dial(misdn/g:Intern/75,30)
exten => 76,2,Hangup
exten => 76,103,Playtones(busy)
exten => 76,104,Wait(2)
exten => 76,105,Hangup(17)
;--------------------------------------------------------------
; fuer alle dialmakro nach draussen, mit runfnummernunterdrueckung
exten => _XX.,1,NoOp(${CALLERIDNUM})
exten => _XX.,2,Set(foo=${DB_EXISTS(Funktionen/CLIP-${CALLERIDNUM})})
exten => _XX.,3,GotoIf(${DB_EXISTS(Funktionen/CLIP-${CALLERIDNUM})}?50:60) ; 50 wenn
anonym, 60 wenn normal rauswahl (clip)
exten => _XX.,50,SetCallerPres(prohib_not_screened)
exten => _XX.,51,Goto(102)
exten => _XX.,60,SetCallerid(325444${CALLERIDNUM})
exten => _XX.,61,Goto(102)
exten => _XX.,102,Dial(misdn/g:Anlagenanschluss/${EXTEN},50,WT)
exten => _XX.,103,Hangup
; if the called party is busy
exten => _XX.,203,Playtones(busy)
exten => _XX.,204,Wait(10)
exten => _XX.,205,Hangup
; if all zaptel channels in that group are in use
; or the D channels are down
exten => _XX.,303,Playtones(congestion)
exten => _XX.,304,Wait(10)
exten => _XX.,305,Hangup
;Callpickup, die blinkende Taste druecken
exten => _*8.,1,PickUpChan(SIP/${EXTEN:2})
exten => i,1,Congestion
exten => h,1,NoOp(${EXTEN}) ; weiter, es muss geguckt werden ob der anruf auf die 75 ging
dann devstate
exten => t,1,Busy
extensions.conf
Page 4 of 4
exten => h,2,DevState(75,0) ; lights off!
exten => o,1,Goto(zentrale_ivr,s,1)
[zentrale]
; context fuer zentrale
exten => s,1,Dial(SIP/10&SIP/20&SIP/30&SIP/40&SIP/60&SIP/70,50,wWt)
exten => s,2,Hangup
exten => s,102,Playtones(busy)
exten => s,103,Wait(2)
exten => s,104,Hangup(

Hoffe mir kann jemand helfen