Hallo!
Ich habe ein Problem mit meinem Trixboxserver. Also ich habe mich schon durchgearbeitet durch diverse Tutorials und es auch soweit geschafft, dass ich von meinem PC-Softphone ein internes ISDN-Telefon anrufen kann und auch normal telefonieren kann. Sobald ich aber den Hörer des ISDN-Fons abhebe oder gar versuche zu wählen, kommt nix. Weder Freizeichen noch der Wahlvorgang selbst
Also ich bräuchte dringend Hilfe, sodass ich die ISDN-Fons zum anwählen bzw Gesprächsaufbau verwenden kann.
Im Anschluss ein paar configs sowie die Ausgabe von Asterisk, wenn ich den Hörer am ISDN-Telefon abhebe.
ASTERISK DEBUG OUTPUT - ISDN
/etc/misdn-init.conf
/etc/asterisk/misdn.conf
Ich habe ein Problem mit meinem Trixboxserver. Also ich habe mich schon durchgearbeitet durch diverse Tutorials und es auch soweit geschafft, dass ich von meinem PC-Softphone ein internes ISDN-Telefon anrufen kann und auch normal telefonieren kann. Sobald ich aber den Hörer des ISDN-Fons abhebe oder gar versuche zu wählen, kommt nix. Weder Freizeichen noch der Wahlvorgang selbst
Also ich bräuchte dringend Hilfe, sodass ich die ISDN-Fons zum anwählen bzw Gesprächsaufbau verwenden kann.
Im Anschluss ein paar configs sowie die Ausgabe von Asterisk, wenn ich den Hörer am ISDN-Telefon abhebe.
ASTERISK DEBUG OUTPUT - ISDN
Code:
P[ 1] % GOT L2 Activate Info.
P[ 1] channel with stid:0 not in use!
P[ 1] --> new_process: New L3Id: 10042
P[ 1] set_channel: bc->channel:0 channel:-1
P[ 1] NO USERUESRINFO
P[ 1] --> found chan: 1
P[ 1] set_chan_in_stack: 1
P[ 1] I IND :SETUP oad:106 dad: pid:11 state:none
P[ 1] --> channel:1 mode:NT cause:16 ocause:16 rad: cad:
P[ 1] --> info_dad: onumplan:0 dnumplan: rnumplan: cpnnumplan:0
P[ 1] --> caps:Speech pi:0 keypad: sending_complete:0
P[ 1] --> screen:0 --> pres:1
P[ 1] --> addr:50010102 l3id:10042 b_stid:0 layer_id:50010180
P[ 1] --> facility:Fac_None out_facility:Fac_None
P[ 1] --> bc_state:BCHAN_CLEANED
P[ 1] --> Bearer: Speech
P[ 1] --> Codec: Alaw
P[ 0] --> * NEW CHANNEL dad: oad:106
P[ 1] read_config: Getting Config
P[ 1] --> CTON: Unknown
P[ 1] --> EXPORT_PID: pid:11
P[ 1] --> PRES: Restricted (1)
P[ 1] --> SCREEN: Unscreened (0)
P[ 1] * Queuing chan 0xa0bca80
P[ 1] ph_control: c1:2310 c2:12
P[ 1] I SEND:RELEASE_COMPLETE oad:106 dad: pid:11
P[ 1] --> bc_state:BCHAN_CLEANED
P[ 1] --> channel:1 mode:NT cause:16 ocause:1 rad: cad:
P[ 1] --> info_dad: onumplan:0 dnumplan: rnumplan: cpnnumplan:0
P[ 1] --> caps:Speech pi:0 keypad: sending_complete:0
P[ 1] --> screen:0 --> pres:1
P[ 1] --> addr:50010102 l3id:10042 b_stid:0 layer_id:50010180
P[ 1] --> facility:Fac_None out_facility:Fac_None
P[ 1] --> CC_RELEASE_CR: Faking Release_cr for 41000101 l3id:10042
P[ 1] --> lib: RELEASE_CR Ind with l3id:10042
P[ 1] --> lib: CLEANING UP l3id: 10042
P[ 1] --> hangup
P[ 1] * IND : HANGUP pid:11 ctx:intern dad: oad:106 State:EXTCANTMATCH
P[ 1] --> l3id:10042
P[ 1] --> cause:16
P[ 1] --> out_cause:16
P[ 1] --> Channel: mISDN/1-u9 hungup new state:CLEANING
P[ 1] $$$ CLEANUP CALLED pid:11
P[ 1] empty_chan_in_stack: 1
P[ 0] handle_bchan: BC not found for prim:281 with addr:51010102 dinfo:0
P[ 0] handle_bchan: BC not found for prim:120281 with addr:51010102 dinfo:0
P[ 1] % GOT L2 DeActivate Info./etc/misdn-init.conf
Code:
#
# 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,hfcpci
#
# 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
#
nt_ptmp=1
#
# 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/etc/asterisk/misdn.conf
Code:
;
; chan_misdn sample config
;
; general section:
;
; for debugging and general setup, things that are not bound to port groups
;
[general]
;
; Sets the Path to the misdn-init.conf (for nt_ptp mode checking)
;
misdn_init=/etc/misdn-init.conf
; set debugging flag:
; 0 - No Debug
; 1 - mISDN Messages and * - Messages, and * - State changes
; 2 - Messages + Message specific Informations (e.g. bearer capability)
; 3 - very Verbose, the above + lots of Driver specific infos
; 4 - even more Verbose than 3
;
; default value: 0
;
debug=4
; set debugging file and flags for mISDNuser (NT-Stack)
;
; flags can be or'ed with the following values:
;
; DBGM_NET 0x00000001
; DBGM_MSG 0x00000002
; DBGM_FSM 0x00000004
; DBGM_TEI 0x00000010
; DBGM_L2 0x00000020
; DBGM_L3 0x00000040
; DBGM_L3DATA 0x00000080
; DBGM_BC 0x00000100
; DBGM_TONE 0x00000200
; DBGM_BCDATA 0x00000400
; DBGM_MAN 0x00001000
; DBGM_APPL 0x00002000
; DBGM_ISDN 0x00004000
; DBGM_SOCK 0x00010000
; DBGM_CONN 0x00020000
; DBGM_CDATA 0x00040000
; DBGM_DDATA 0x00080000
; DBGM_SOUND 0x00100000
; DBGM_SDATA 0x00200000
; DBGM_TOPLEVEL 0x40000000
; DBGM_ALL 0xffffffff
;
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
; some pbx systems do cut the L1 for some milliseconds, to avoid
; dropping running calls, we can set this flag to yes and tell
; mISDNuser not to drop the calls on L2_RELEASE
ntkeepcalls=no
; the big trace
;
; default value: [not set]
;
;tracefile=/var/log/asterisk/misdn.log
; set to yes if you want mISDN_dsp to bridge the calls in HW
;
; default value: yes
;
bridging=no
;
; watches the L1s of every port. If one l1 is down it tries to
; get it up. The timeout is given in seconds. with 0 as value it
; does not watch the l1 at all
;
; default value: 0
;
; this option is only read at loading time of chan_misdn,
; which means you need to unload and load chan_misdn to change the
; value, an asterisk restart should do the trick
;
l1watcher_timeout=0
; stops dialtone after getting first digit on nt Port
;
; default value: yes
;
stop_tone_after_first_digit=yes
; whether to append overlapdialed Digits to Extension or not
;
; default value: yes
;
append_digits2exten=yes
;;; CRYPTION STUFF
; Whether to look for dynamic crypting attempt
;
; default value: no
;
dynamic_crypt=no
; crypt_prefix, what is used for crypting Protocol
;
; default value: [not set]
;
crypt_prefix=**
; Keys for cryption, you reference them in the dialplan
; later also in dynamic encr.
;
; default value: [not set]
;
crypt_keys=test,muh
; users sections:
;
; name your sections as you which but not "general" !
; the sections are Groups, you can dial out in extensions.conf
; with Dial(mISDN/g:extern/101) where extern is a section name,
; chan_misdn tries every port in this section to find a
; new free channel
;
; The default section is not a group section, it just contains config elements
; which are inherited by group sections.
;
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP 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 SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; channel. Defaults to "no".
; 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 SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
; The option represents the number of milliseconds by which the new
; jitter buffer will pad its size. the default is 40, so without
; modification, the new jitter buffer will set its size to the jitter
; value plus 40 milliseconds. increasing this value may help if your
; network normally has low jitter, but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
[default]
; define your default context here
;
; default value: default
;
context=misdn
; language
;
; default value: en
;
language=en
;
; sets the musiconhold class
;
musicclass=default
;
; Either if we should produce DTMF Tones ourselves
;
senddtmf=yes
;
; If we should generate Ringing for chan_sip and others
;
far_alerting=no
;
; Here you can list which bearer capabilities should be allowed:
; all - allow any bearer capability
; speech - allow speech
; 3_1khz - allow 3.1KHz audio
; digital_unrestricted - allow unrestricted digital
; digital_restricted - allow restricted digital
; video - allow video
;
; Example:
; allowed_bearers=speech,3_1khz
;
allowed_bearers=all
; Prefixes for national and international, those are put before the
; oad if an according dialplan is set by the other end.
;
; default values: nationalprefix : 0
; internationalprefix : 00
;
nationalprefix=0
internationalprefix=00
; set rx/tx gains between -8 and 8 to change the RX/TX Gain
;
; default values: rxgain: 0
; txgain: 0
;
rxgain=0
txgain=0
; some telcos especially in NL seem to need this set to yes, also in
; switzerland this seems to be important
;
; default value: no
;
te_choose_channel=no
;
; This option defines, if chan_misdn should check the L1 on a PMP
; before making a group call on it. The L1 may go down for PMP Ports
; so we might need this.
; But be aware! a broken or plugged off cable might be used for a group call
; as well, since chan_misdn has no chance to distinguish if the L1 is down
; because of a lost Link or because the Provider shut it down...
;
; default: no
;
pmp_l1_check=no
;
; in PMP this option defines which cause should be sent out to
; the 3. caller. chan_misdn does not support callwaiting on TE
; PMP side. This allows to modify the RELEASE_COMPLETE cause
; at least.
;
reject_cause=16
;
; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING),
; this requests additional Infos, so we can waitfordigits
; without much issues. This works only for PTP Ports
;
; default value: no
;
need_more_infos=no
;
; set this to yes if you want to disconnect calls when a timeout occurs
; for example during the overlapdial phase
;
nttimeout=no
; Set the method to use for channel selection:
; standard - Use the first free channel starting from the lowest number.
; standard_dec - Use the first free channel starting from the highest number.
; round_robin - Use the round robin algorithm to select a channel. Use this
; if you want to balance your load.
;
; default value: standard
;
method=standard
; specify if chan_misdn should collect digits before going into the
; dialplan, you can choose yes=4 Seconds, no, or specify the amount
; of seconds you need;
;
overlapdial=yes
;
; dialplan means Type Of Number in ISDN Terms (for outgoing calls)
;
; there are different types of the dialplan:
;
; dialplan -> outgoing Number
; localdialplan -> callerid
; cpndialplan -> connected party number
;
; dialplan options:
;
; 0 - unknown
; 1 - International
; 2 - National
; 4 - Subscriber
;
; This setting is used for outgoing calls
;
; default value: 0
;
dialplan=0
localdialplan=0
cpndialplan=0
;
; turn this to no if you don't mind correct handling of Progress Indicators
;
early_bconnect=yes
;
; turn this on if you like to send Tone Indications to a Incoming
; isdn channel on a TE Port. Rarely used, only if the Telco allows
; you to send indications by yourself, normally the Telco sends the
; indications to the remote party.
;
; default: no
;
incoming_early_audio=no
; uncomment the following to get into s extension at extension conf
; there you can use DigitTimeout if you can't or don't want to use
; isdn overlap dial.
; note: This will jump into the s exten for every exten!
;
; default value: no
;
;always_immediate=no
;
; set this to yes if you want to generate your own dialtone
; with always_immediate=yes, else chan_misdn generates the dialtone
;
; default value: no
;
nodialtone=no
; uncomment the following if you want callers which called exactly the
; base number (so no extension is set) jump to the s extension.
; if the user dials something more it jumps to the correct extension
; instead
;
; default value: no
;
;immediate=no
; uncomment the following to have hold and retrieve support
;
; default value: no
;
;hold_allowed=yes
; Pickup and Callgroup
;
; default values: not set = 0
; range: 0-63
;
;callgroup=1
;pickupgroup=1
;
; these are the exact isdn screening and presentation indicators
; if -1 is given for either value the presentation indicators are used
; from asterisks SetCallerPres application.
; s=0, p=0 -> callerid presented
; s=1, p=1 -> callerid restricted (the remote end does not see it!)
;
; default values s=-1, p=-1
presentation=-1
screen=-1
; This enables echo cancellation with the given number of taps.
; Be aware: Move this setting only to outgoing portgroups!
; A value of zero turns echo cancellation off.
;
; possible values are: 0,32,64,128,256,yes(=128),no(=0)
;
; default value: no
;
;echocancel=no
; Set this to no to disable echotraining. You can enter a number > 10
; the value is a multiple of 0.125 ms.
;
; default value: no
; yes = 2000
; no = 0
;
echotraining=no
;
; chan_misdns jitterbuffer, default 4000
;
jitterbuffer=4000
;
; change this threshold to enable dejitter functionality
;
jitterbuffer_upper_threshold=0
;
; change this to yes, if you want to bridge a mISDN data channel to
; another channel type or to an application.
;
hdlc=no
;
; defines the maximum amount of incoming calls per port for
; this group. Calls which exceed the maximum will be marked with
; the channel variable MAX_OVERFLOW. It will contain the amount of
; overflowed calls
;
max_incoming=-1
;
; defines the maximum amount of outgoing calls per port for this group
; exceeding calls will be rejected
;
max_outgoing=-1
[intern]
; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
ports=1
; context where to go to when incoming Call on one of the above ports
context=intern
msns=*
;[internPP]
;
; adding the postfix 'ptp' to a port number is obsolete now, chan_misdn
; parses /etc/misdn-init.conf and sets the ptp mode to the corresponding
; configs. For backwards compatibility you can still set ptp here.
;
;ports=3
;[first_extern]
; again port defs
;ports=4
; again a context for incoming calls
;context=Extern1
; msns for te ports, listen on those numbers on the above ports, and
; indicate the incoming calls to asterisk
; here you can give a comma separated list or simply an '*' for
; any msn.
;msns=*
; here an example with given msns
;[second_extern]
;ports=5
;context=Extern2
;callerid=15
;msns=102,144,101,104
