app_capiCD.so kann nicht geladen werden (bristuff0.3.0)

[the_brain-one]

Hallo,

ich habe eine SuSe Linux 10.0 mit einem Asterisk1.2.4 installiert. Soweit konnte ich meine Grundkonfiguration ohne Probleme konfigurieren.

Da ich eine Karte mit HFC Chip habe (Gerdes Primux 4S0-NT) habe ich die bristuff-Patches installiert. Die Konfigurationen habe ich aus den Tutorials der IX's genommen.
Seit ich aber bristuff installiert habe startet mein Asterisk nicht mehr und bringt mir folgende Meldungen. Ich hab jetzt mal nur den Teil ausgeschnitten, der Relevant ist:

== Parsing '/etc/asterisk/musiconhold.conf': Found
Mar 2 10:12:02 WARNING[5526]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing... Sound may be choppy.
[pbx_dundi.so] => (Distributed Universal Number Discovery (DUNDi))
== Parsing '/etc/asterisk/dundi.conf': Found
== Using TOS bits 0
== DUNDi Ready and Listening on 0.0.0.0 port 4520
== Registered application 'DUNDiLookup'
== Registered custom function DUNDILOOKUP
[pbx_functions.so] => (Builtin dialplan functions)
== Registered custom function MD5
== Registered custom function CHECK_MD5
== Registered custom function MATH
== Registered custom function GROUP_COUNT
== Registered custom function GROUP_MATCH_COUNT
== Registered custom function GROUP
== Registered custom function GROUP_LIST
== Registered custom function FIELDQTY
== Registered custom function REGEX
== Registered custom function LEN
== Registered custom function STRFTIME
== Registered custom function EVAL
== Registered custom function CDR
== Registered custom function ISNULL
== Registered custom function SET
== Registered custom function EXISTS
== Registered custom function IF
== Registered custom function IFTIME
== Registered custom function ENV
== Registered custom function DB
== Registered custom function DB_EXISTS
== Registered custom function TIMEOUT
Warning, flexible rate not heavily tested!
== Registered custom function LANGUAGE
== Registered custom function MUSICCLASS
[pbx_ael.so] => (Asterisk Extension Language Compiler)
-- Registered extension context 'macro-std-exten-ael'
-- Added extension 's' priority 1 to macro-std-exten-ael
-- Added extension 's' priority 2 to macro-std-exten-ael
-- Added extension 's' priority 3 to macro-std-exten-ael
-- Added extension 's' priority 4 to macro-std-exten-ael
-- Added extension 's' priority 5 to macro-std-exten-ael
-- Added extension 'sw-4-BUSY' priority 1 to macro-std-exten-ael
-- Added extension 'sw-4-BUSY' priority 2 to macro-std-exten-ael
-- Added extension '_sw-4-.' priority 1 to macro-std-exten-ael
-- Added extension 'a' priority 1 to macro-std-exten-ael
-- Added extension 'a' priority 2 to macro-std-exten-ael
-- Registered extension context 'ael-demo'
-- Added extension 's' priority 1 to ael-demo
-- Added extension 's' priority 2 to ael-demo
-- Added extension 's' priority 3 to ael-demo
-- Added extension 's' priority 4 to ael-demo
-- Added extension 's' priority 5 to ael-demo
-- Added extension 's' priority 6 to ael-demo
-- Added extension 's' priority 8 to ael-demo
-- Added extension 's' priority 9 to ael-demo
-- Added extension 's' priority 10 to ael-demo
-- Added extension 's' priority 11 to ael-demo
-- Added extension 's' priority 12 to ael-demo
-- Added extension 's' priority 7 to ael-demo
-- Added extension '2' priority 1 to ael-demo
-- Added extension '2' priority 2 to ael-demo
-- Added extension '3' priority 1 to ael-demo
-- Added extension '3' priority 2 to ael-demo
-- Added extension '500' priority 1 to ael-demo
-- Added extension '500' priority 2 to ael-demo
-- Added extension '500' priority 3 to ael-demo
-- Added extension '500' priority 4 to ael-demo
-- Added extension '600' priority 1 to ael-demo
-- Added extension '600' priority 2 to ael-demo
-- Added extension '600' priority 3 to ael-demo
-- Added extension '600' priority 4 to ael-demo
-- Added extension '_1234' priority 1 to ael-demo
-- Added extension '#' priority 1 to ael-demo
-- Added extension '#' priority 2 to ael-demo
-- Added extension 't' priority 1 to ael-demo
-- Added extension 'i' priority 1 to ael-demo
[pbx_spool.so] => (Outgoing Spool Support)
[pbx_loopback.so] => (Loopback Switch)
[pbx_config.so] => (Text Extension Configuration)
== Parsing '/etc/asterisk/extensions.conf': Found
== Setting global variable 'RINGTIME' to '30'
-- Registered extension context 'default'
-- Added extension '_ZXX' priority 1 to default
-- Added extension '_4ZXX' priority 1 to default
-- Registered extension context 'macro-anrufintern'
-- Added extension 's' priority 1 to macro-anrufintern
-- Added extension 's' priority 2 to macro-anrufintern
-- Added extension 's-NOANSWER' priority 1 to macro-anrufintern
-- Added extension 's-BUSY' priority 1 to macro-anrufintern
-- Added extension '_s-.' priority 1 to macro-anrufintern
-- Registered extension context 'local'
-- Including context 'default' in context 'local'
[pbx_realtime.so] => (Realtime Switch)
[chan_mgcp.so] => (Media Gateway Control Protocol (MGCP))
== Parsing '/etc/asterisk/mgcp.conf': Found
== MGCP Listening on 0.0.0.0:2727
== Using TOS bits 0
== Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Automatically generated pseudo channel
== Registered channel type 'Zap' (Zapata Telephony Driver w/PRI)
== Manager registered action ZapTransfer
== Manager registered action ZapHangup
== Manager registered action ZapDialOffhook
== Manager registered action ZapDNDon
== Manager registered action ZapDNDoff
== Manager registered action ZapShowChannels
== Registered application 'zapEC'
== Registered application 'zapCD'
[chan_local.so] => (Local Proxy Channel)
== Registered channel type 'Local' (Local Proxy Channel Driver)
[chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
== Registered custom function IAXPEER
Mar 2 10:12:03 WARNING[5526]: chan_iax2.c:9991 load_module: Unable to open IAX timing interface: No such file or directory
Mar 2 10:12:03 NOTICE[5526]: chan_iax2.c:10015 load_module: servername = linux
== Registered application 'IAX2Provision'
== Manager registered action IAXpeers
== Manager registered action IAXnetstats
== Parsing '/etc/asterisk/iax.conf': Found
-- doing lookup for '216.207.245.47'
== Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
== Using TOS bits 16
== Binding IAX2 to default address 0.0.0.0:4569
== IAX Ready and Listening
== Loaded firmware 'iaxy.bin'
== Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
[chan_agent.so] => (Agent Proxy Channel)
== Registered channel type 'Agent' (Call Agent Proxy Channel)
== Registered application 'AgentLogin'
== Registered application 'AgentCallbackLogin'
== Registered application 'AgentMonitorOutgoing'
== Manager registered action Agents
== Manager registered action AgentLogoff
== Manager registered action AgentCallbackLogin
== Parsing '/etc/asterisk/agents.conf': Found
[chan_capi.so] => (Common ISDN API for Asterisk)
Mar 2 10:12:03 ERROR[5526]: chan_capi.c:2622 load_module: Unable to load config capi.conf, CAPI disabled
[chan_phone.so] => (Linux Telephony API Support)
== Parsing '/etc/asterisk/phone.conf': Found
== Registered channel type 'Phone' (Standard Linux Telephony API Driver)
[chan_skinny.so] => (Skinny Client Control Protocol (Skinny))
== Parsing '/etc/asterisk/skinny.conf': Found
== Skinny listening on 0.0.0.0:2000
== Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny))
[chan_features.so] => (Feature Proxy Channel)
== Registered channel type 'Feature' (Feature Proxy Channel Driver)
[chan_oss.so] => (OSS Console Channel Driver)
== Parsing '/etc/asterisk/oss.conf': Found
== Registered channel type 'Console' (OSS Console Channel Driver)
[chan_sip.so] => (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
-- SIP Seeding peer from astdb: '100' at [email protected]:5060 for 120
== SIP Listening on 0.0.0.0:5060
== Using TOS bits 0
== Parsing '/etc/asterisk/sip_notify.conf': Found
== Registered channel type 'SIP' (Session Initiation Protocol (SIP))
== Registered application 'SIPDtmfMode'
== Registered application 'SIPAddHeader'
== Registered application 'SIPGetHeader'
== Registered custom function SIP_HEADER
== Registered custom function SIPPEER
== Registered custom function SIPCHANINFO
== Registered custom function CHECKSIPDOMAIN
== Manager registered action SIPpeers
== Manager registered action SIPshowpeer
== Manager registered action SIPNotify
[app_sms.so] => (SMS/PSTN handler)
== Registered application 'SMS'
[cdr_custom.so] => (Customizable Comma Separated Values CDR Backend)
== Parsing '/etc/asterisk/cdr_custom.conf': Found
[app_hasnewvoicemail.so] => (Indicator for whether a voice mailbox has messages in a given folder.
== Registered custom function VMCOUNT
== Registered application 'HasVoicemail'
== Registered application 'HasNewVoicemail'
[format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM))
== Registered file format wav49, extension(s) WAV|wav49
[app_url.so] => (Send URL Applications)
== Registered application 'SendURL'
[app_test.so] => (Interface Test Application)
== Registered application 'TestClient'
== Registered application 'TestServer'
[app_eval.so] => (Reevaluates strings)
== Registered application 'Eval'
[app_chanspy.so] => (Listen to the audio of an active channel
)
== Registered application 'ChanSpy'
[app_sendtext.so] => (Send Text Applications)
== Registered application 'SendText'
[app_exec.so] => (Executes applications)
== Registered application 'Exec'
[app_txtcidname.so] => (TXTCIDName)
== Registered application 'TXTCIDName'
[cdr_manager.so] => (Asterisk Call Manager CDR Backend)
== Parsing '/etc/asterisk/cdr_manager.conf': Found
[app_capiCD.so]Mar 2 10:12:03 WARNING[5526]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: capidebug
Mar 2 10:12:03 WARNING[5526]: loader.c:554 load_modules: Loading module app_capiCD.so failed!


Hat jemand eine Idee, wie ich dieses Problem beheben kann?? In Sachen Asterisk bin ich noch ziemlich unerfahren.

 

[the_brain-one]

Ich hab den Fehler gefunden.

Wenn ich in der modules.conf das app_capiCD.so nicht lade, startet mein Asterisk ohne Fehler.

Die Frage ist jetzt nur brauch ich dieses Modul und wofür ist es da??

 

[cibi]

Das wird von den alten Capitreibern benötigt, die mittlerweile mit bristuff installiert werden. s. auch
http://www.ip-phone-forum.de/showthread.php?t=81433
Diese werden allerdings nur installiert wenn auch ein CAPI Treiber vorhanden ist.
Deswegen die Frage:
Funktioniert die Gerdes Karte mit den bristuff Treibern (qozap) ?
(solnage die capi treiber geladen sind nämlich nicht)


PS/OT: Hast du es mal mit ohne bristuff, mit den capitreiber und chan_capi_cm probiert?

 

[the_brain-one]

Welchen Capi-Treiber meinst Du jetzt. Den, der mit der Karte mit installiert wird??

Also mit chan_capi bekomme ich eine Fehlermeldung, dass das Modul chan_capi nicht geladen werden kann, weil capi nicht gestartet ist.

 

[rajo]

Solange es eine Karte mit HFC-Chipsatz ist, wir bristuff und *nur* Bristuff dafür benötigt. Kein Capi, kein Hisax, kein sonstwas.

Chan-capi kann bei HFC-basierten karten komplett weg.

 

[cibi]

Solange es eine Karte mit HFC-Chipsatz ist, wir bristuff und *nur* Bristuff dafür benötigt. Kein Capi, kein Hisax, kein sonstwas.

Chan-capi kann bei HFC-basierten karten komplett weg.

Die Gerdes Karten bringen (vermutlich) einen Capi Treiber mit sich:
http://www.ip-phone-forum.de/showthread.php?t=81433

Auch habe diese einen DSP Chip, den ich zumindest bei den Abbildungen der andren Karten nicht sehe.

Kann also sein, das die Karte mit bristuff läuft, aber eben nicht alle Eigenschaften der Karte unterstützt werden.

Vielleicht kann jemand, der Erfahrung mit den Mehrfachkarten hat, dazu was sagen.

 

[the_brain-one]

Die Karte bringt auf auf jedenfall einen capi-Treiber mit. In einer grafischen Oberfläche kann ich die Ports auf NT oder TE umstellen. Zusätzlich muss ich das hardwaremässig noch mal auf der Karte durch "Jumper" machen. Ach ja, die haben tatsächlich diesen DSP-Chip drauf und einen HFC-4S-Chip.
Hat jemand eine Ahnung was ich jetzt brauche und welche Konfigurationsdateien ich dann wie zu konfigurieren habe.

 

[the_brain-one]

Brauch ich den florz-Patch eigentlich für Multi-S0 Karten?
Oder brauch ich den nur, wenn ich wirklich zwei sepparate Karten einbauen will?

 

[the_brain-one]

Ich habe jetzt mal versucht, die zapata.conf anzupassen (NT und TE).

Jetzt bekomme ich folgende Fehlermeldungen:

[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Mar 2 15:46:29 WARNING[5741]: chan_zap.c:1069 zt_open: Unable to open '/dev/zap/channel': No such file or directory
Mar 2 15:46:29 ERROR[5741]: chan_zap.c:7104 mkintf: Unable to open channel 1: No such file or directory
here = 0, tmp->channel = 1, channel = 1
Mar 2 15:46:29 ERROR[5741]: chan_zap.c:11405 setup_zap: Unable to register channel '1-2'
Mar 2 15:46:29 WARNING[5741]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1
Mar 2 15:46:29 WARNING[5741]: loader.c:499 load_modules: Loading module chan_zap.so failed!
linux:~ # Warning, flexible rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe


Meine zapata.conf sieht so aus:

[channels]

; Karte für extern S0
switchtype=euroisdn
signalling=bri_cpe_ptmp
pridialplan=local
prilocaldialplan=local
usecallerid=yes
overlapdial=no
immidiate=no
group=1
context=tcom-trunk
channel=>1-2

; Karte für intern NT
switchtype=euroisdn
signalling=bri_net_ptmp
pridialplan=local
prilocaldialplan=local
usecallerid=yes
overlapdial=no
immidiate=no
group=2
context=pbx-trunk
channel=>4-5


Hat jemand eine Idee?