[chan-capi-users] chan_capi trunk not compiling with Asterisk trunk SVN-trunk-r240667M

Thu Mar 4 17:21:55 CET 2010

> Please try new svn trunk of chan_capi. I just made some 
> changes for current asterisk trunk.
> 
> But I did not test the changes. Since I just fixed the API, 
> some changes in asterisk for functionality may cause problems 
> and need changes in chan_capi too.
> 
> Armin

Hi,

It seems that my success message was a bit premature. Receiving calls
from the PSTN works fine, but calling out doesn't work. The telephone
rings, but upon picking up asterisk crashes. I hope you can get some
clues from my attached output, confs and backtrace (contains bt and bt
full):

Asterisk console output
-----------------------

*CLI> core set verbose 4
Verbosity was 0 and is now 4
*CLI>
*CLI> capi debug
CAPI Message Debugging Enabled
*CLI>

*CLI>   == Using UDPTL CoS mark 5
    -- Executing [01234123456 at default:1] Dial("SIP/1500-00000000",
"CAPI/ISDN1/01234123456") in new stack
       > data = ISDN1/01234123456 format=8
       > parsed dialstring: 'ISDN1' 'NULL' '01234123456' ''
       > capi request for interface 'ISDN1'
  == ISDN1#02: setting format alaw - 0x8 (alaw)
       > parsed dialstring: 'ISDN1' 'NULL' '01234123456' ''
       > capi: peerlink -1 allocated, peer is unlinked
  == ISDN1#02: Call CAPI/ISDN1#02/01234123456-0   (pres=0x00, ton=0x00)
    -- ISDN1#02: received CONNECT_CONF PLCI = 0x101
       > chan_capi devicestate requested for ISDN1#02/01234123456 is 'In
use'
    -- Called ISDN1/01234123456
    -- ISDN1#02: info element CALL PROCEEDING
    -- chan_capi queue frame: TYPE: Control (4) SUBCLASS: Unknown
control '15' (15) ] [ISDN1#02]
    -- ISDN1#02: info element CHANNEL IDENTIFICATION 89
    -- CAPI/ISDN1#02/01234123456-0 is proceeding passing it to
SIP/1500-00000000
    -- ISDN1#02: info element ALERTING
    -- chan_capi queue frame: TYPE: Control (4) SUBCLASS: Unknown
control '14' (14) ] [ISDN1#02]
    -- chan_capi queue frame: TYPE: Control (4) SUBCLASS: Ringing (3) ]
[ISDN1#02]
    -- ISDN1#02: info element PI 82 88
       > chan_capi devicestate requested for ISDN1#02/01234123456 is 'In
use'
    -- CAPI/ISDN1#02/01234123456-0 is making progress passing it to
SIP/1500-00000000
    -- CAPI/ISDN1#02/01234123456-0 is ringing
    -- ISDN1#02: info element PI 81 82
    -- ISDN1#02: info element Date/Time 10/03/04 13:17
Speicherzugriffsfehler
srvpbx:~#

extensions.conf
---------------

[default]
;isdn-out
exten => _0.,1,Dial(CAPI/ISDN1/${EXTEN:0})

[isdn-in]
exten => 123456,1,Dial(SIP/1501)   ;MSN anonymised

capi.conf
---------

;
; CAPI config
;
;

; general section

[general]
nationalprefix=0        ; or for example "+49"
internationalprefix=00  ; or for example "+"
;subscriberprefix=+4969 ; prefix including area code (some lines need
this)
rxgain=1.0       ;linear receive gain (1.0 = no change)
txgain=1.0       ;linear transmit gain (1.0 = no change)
language=de      ;set default language
;ulaw=yes        ;set this, if you live in u-law world instead of a-law

;jb.....         ;with Asterisk 1.4 you can configure jitterbuffer,
                 ;see Asterisk documentation for all jb* setting
available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when
placed on hold.

; interface sections ...

[ISDN1]          ;this example interface gets name 'ISDN1' and may be
any
                 ;name not starting with 'g' or 'contr'.
                 ;Use one interface section for each ISDN port!
;ntmode=yes      ;if the ISDN card operates in NT-mode, set this to
'yes'
isdnmode=msn     ;'MSN' (point-to-multipoint) or 'DID' (direct inward
dial)
                 ;when using NT-mode, 'DID' should be set in any case
incomingmsn=*    ;allow incoming calls to this list of MSNs/DIDs, * =
any
;defaultcid=123  ;set a default caller ID to that interface for
dial-out,
                 ;this caller ID will be used when the dial option 'd'
is set.
;controller=0    ;ISDN4BSD default
;controller=7    ;ISDN4BSD USB default
controller=1     ;CAPI controller number of this interface/port
group=1          ;dialout group
;prefix=0        ;set a prefix to the calling number on incoming calls
softdtmf=on      ;enable/disable software DTMF detection, recommended
for AVM cards
relaxdtmf=on     ;in addition to softdtmf, you can use relaxed DTMF
detection
faxdetect=off    ;enable faxdetection and redirection to EXTEN 'fax' for
incoming and/or
                 ;outgoing calls. (default='off', possible values:
'incoming','outgoing','both')
faxdetecttime=0  ;Only detect faxes during the first 'n' seconds of the
call.
                 ;(default '0' meaning for the whole duration of the
call)
accountcode=     ;PBX accountcode to use in CDRs
;amaflags=default;AMA flags for CDR ('default', 'omit', 'billing', or
'documentation')
context=isdn-in  ;context for incoming calls
;holdtype=hold   ;when the PBX puts the call on hold, ISDN HOLD will be
used. If
                 ;set to 'local' (default value), no hold is done and
the PBX may
                 ;play MOH.
;immediate=yes   ;DID: immediate start of PBX with extension 's' if no
digits were
                 ;     received on incoming call (no destination number
yet)
                 ;MSN: start PBX on CONNECT_IND and do not wait for
SETUP/SENDING-COMPLETE.
                 ;     info like REDIRECTINGNUMBER may be lost, but this
is necessary for
                 ;     drivers/pbx/telco which does not send SETUP or
SENDING-COMPLETE.
;echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression. Disable it before
you start recording voicemail
                 ;or your files may get choppy. (you can use
capicommand(echosquelch|no) for this)
;echocancel=yes  ;Dialogic(R) Diva(R) (CAPI) echo cancellation
(yes=g165)

modules.conf
------------

;
; Asterisk configuration file
;
; Module Loader configuration file
;

[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using 'preload'. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those
configuration
; files are initialized.
;
; An example of loading ODBC support would be:
;preload => res_odbc.so
;preload => res_config_odbc.so
;
; Uncomment the following if you wish to use the Speech Recognition API
;preload => res_speech.so
;
; If you want Asterisk to fail if a module does not load, then use
; the "require" keyword. Asterisk will exit with a status code of 2
; if a required module does not load.
;
; require = chan_sip.so
; If you want you can combine with preload
; preload-require = res_odbc.so
;
; If you want, load the GTK console right away.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
;
load => res_musiconhold.so
;
; Load one of: chan_oss, alsa, or console (portaudio).
; By default, load chan_oss only (automatically).
;
noload => chan_alsa.so
;noload => chan_oss.so
noload => chan_console.so

load => res_features.so
load => chan_capi.so    ; CAPI-Verbindungsaufbau

[global]
chan_capi = yes

Many thanks in advance

Daniel
-------------- next part --------------
An embedded and charset-unspecified text was scrubbed...
Name: gdb.txt
Url: http://lists.melware.net/pipermail/chan-capi-users/attachments/20100304/d07982cb/attachment.txt 