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

[Armin Schindler]

On Thu, 4 Mar 2010, DLeese at LStelcom.com wrote:
>> 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):

I had a quick look at this, but cannot find the cause in chan_capi so far.
As long as this is an asterisk trunk version, I consider it as unstable and
will not put too much effort into this.

Armin


> 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
>