Hallo Gemeinde,
folgende Situation:
ein Asterisk (fli4l) per AVM-Karte an Telekom-ISDN, dahinter mit einer HFC-Karte (gekreuzte Adern) eine Telefonanlage (Auerswald 2206). Zum "bug jagen" habe ich noch ein ISDN-Telefon an die HFC-Karte (parallel zur Anlage) angeschlossen.
Rauswählen über Sipgate und ISDN funktioniert einwandfrei (mit wahl der entsprechenden "Amtkennziffern").
Ankommende ISDN-Anrufe werden an die Anlage weiterübermittelt und dort auch an allen Apparaten angezeigt.
Problem: ankommende Rufe über Sipgate werden nirgendwo signalisiert. Als Anrufer habe ich nicht mal eine Signalisierung des Klingelns; es geht nach einer gewissen Zeit (NICHT sofort) die Mailbox von Sipgate dran.
Wenn ich in der sip.conf die Zeile "context=ankommend" einfüge, höre ich als Anrufer ein klingeln, bei den Telefonen wird nichts signalisiert.
Allerdings erscheint dann auch die Fehlermeldung
*CLI> Jul 12 10:51:24 NOTICE[14017]: pbx.c:1739 pbx_extension_helper: Cannot find extension context 'ankommend'
Meine Konfigurationsdateien:
sip.conf
extension.conf
zapata.conf
Beim starten von Asterisk erscheinen folgende Meldungen:
Anzeige von Astersisk bei einem ISDN-Anruf:
== ISDN1: Incoming call '01762819xxxx -> '742xxx'
-- ISDN1: Updated channel name: CAPI/ISDN1/742xxx-1
-- Executing Dial("CAPI/ISDN1/742606-1", "Zap/g1/742xxx|20|Tt") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/742xxx
-- Zap/1-1 is proceeding passing it to CAPI/ISDN1/742xxx-1
-- Zap/1-1 is ringing
Anzeige von Asterisk bei einem Anruf über SIP:
nichts
Anzeige von Asterisk bei einem Anruf über SIP (mit "context=ankommend"):
*CLI> Jul 12 10:51:24 NOTICE[14017]: pbx.c:1739 pbx_extension_helper: Cannot find extension context 'ankommend'
Anzeige von Asterisk wenn ich telefoniere (über SIP):
*CLI> -- Accepting overlap voice call from '' to '90' on channel 0/2, span 1
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial("Zap/2-1", "SIP/01762819xxxx@sipgate-raisch||Ttr") in new stack
-- Called 01762819xxxx@sipgate-raisch
-- SIP/sipgate-raisch-2444 answered Zap/2-1
-- Channel 0/2, span 1 got hangup request
== Spawn extension (default, 901762819xxxx, 1) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
2. Frage:
Momentan werden die Rufnummern der ISDN-Anrufer an den Telefonen angezeigt; allerdings MIT der "Amtkennziffer". Also wird aus "06073blablubb" ein "006073blablubb"
Ist das irgendwo einstellbar, das die erste Ziffer nicht angezeigt wird?
Vielen Dank schon mal...
ciao
Indy
folgende Situation:
ein Asterisk (fli4l) per AVM-Karte an Telekom-ISDN, dahinter mit einer HFC-Karte (gekreuzte Adern) eine Telefonanlage (Auerswald 2206). Zum "bug jagen" habe ich noch ein ISDN-Telefon an die HFC-Karte (parallel zur Anlage) angeschlossen.
Rauswählen über Sipgate und ISDN funktioniert einwandfrei (mit wahl der entsprechenden "Amtkennziffern").
Ankommende ISDN-Anrufe werden an die Anlage weiterübermittelt und dort auch an allen Apparaten angezeigt.
Problem: ankommende Rufe über Sipgate werden nirgendwo signalisiert. Als Anrufer habe ich nicht mal eine Signalisierung des Klingelns; es geht nach einer gewissen Zeit (NICHT sofort) die Mailbox von Sipgate dran.
Wenn ich in der sip.conf die Zeile "context=ankommend" einfüge, höre ich als Anrufer ein klingeln, bei den Telefonen wird nichts signalisiert.
Allerdings erscheint dann auch die Fehlermeldung
*CLI> Jul 12 10:51:24 NOTICE[14017]: pbx.c:1739 pbx_extension_helper: Cannot find extension context 'ankommend'
Meine Konfigurationsdateien:
sip.conf
Code:
[general]
port = 5060
bindaddr = 0.0.0.0
externhost =
Localnet = 192.168.0.2 255.255.255.0
srvlookup = yes
context = default
disallow=all
allow=gsm
canreinvite=no
tos=0x18
insecure=very
nat=no
dtmfmode=info
register => xxxxxxxx
register => yyyyyyyy
[sipgate-raisch]
type=friend
username=xxxx
secret=yyyy
host=sipgate.de
fromuser=yyyyyy
fromdomain=sipgate.de
context=default
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g726
insecure=very
nat=no
dtmfmode=info
tos=0x18
[sipgate-mohar]
type=friend
username=yxxxxxx
secret=yyyyyyx
host=sipgate.de
fromuser=yxxxxxx
fromdomain=sipgate.de
context=default
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g726
insecure=very
nat=no
dtmfmode=info
tos=0x18
[sipgate_in]
type=peer
fromdomain=sipgate.de
host=sipgate.de
disallow=all
allow=gsm
allow=g726
allow=ulaw
context=ankommend
extension.conf
Code:
[internal]
exten => 742xxx,1,Dial(Zap/g1/742xxx,20,Tt)
exten => 742xxx,2,Voicemail2(u742xxx)
exten => 742xxx,3,Hangup
exten => 742xxx,102,Voicemail2(b742xxx)
exten => 742xxx,103,Hangup
exten => 742xxy,1,Dial(Zap/g1/742xxy,20,Tt)
exten => 742xxy,2,Voicemail2(u742xxy)
exten => 742xxy,3,Hangup
exten => 742xxy,102,Voicemail2(b742xxy)
exten => 742xxy,103,Hangup
exten => 630xxx,1,Dial(Zap/g1/630xxx,20,Tt)
exten => 630xxx,2,Voicemail2(u630xxx)
exten => 630xxx,3,Hangup
exten => 630xxx,102,Voicemail2(b630xxx)
exten => 630xxx,103,Hangup
exten => 630xxy,1,Dial(Zap/g1/630xxy,20,Tt)
exten => 630xxy,2,Voicemail2(u630xxy)
exten => 630xxy,3,Hangup
exten => 630xxy,102,Voicemail2(b630xxy)
exten => 630xxy,103,Hangup
[external-ISDN]
exten => _0.,1,Set(CALLERID(number)=742xxx))
exten => _0.,2,Dial(CAPI/contr1/${EXTEN:1},,Tt)
exten => _0.,3,Congestion
[external-SIP]
exten => _8.,1,Dial(SIP/${EXTEN:1}@sipgate-mohar,45,Ttr)
exten => _8.,2,Congestion
exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate-raisch,,Ttr)
exten => _9.,2,Congestion
[vmailbox]
exten => 8000,1,VoicemailMain
[default]
include => vmailbox
include => internal
; include => external-X100P
include => external-ISDN
include => external-SIP
zapata.conf
Code:
[channels]
language=de
switchtype=euroisdn
signalling=bri_net_ptmp
pridialplan=local
echocancel=yes
immediate=no
;setcallerid(<${CALLERIDNUM}>)
overlapdial=yes
group=1
context=default
channel=>1-2
;
;signalling = bri_cpe_ptmp
;group = 2
;channel => 4-5
Beim starten von Asterisk erscheinen folgende Meldungen:
Code:
fli4l 3.0.1 # asterisk -vvvvvgc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1o, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
== Parsing '/etc/asterisk/dnsmgr.conf': Found
Asterisk Dynamic Loader loading preload modules:
== Parsing '/etc/asterisk/modules.conf': Found
== Manager registered action Ping
== Manager registered action Events
== Manager registered action Logoff
== Manager registered action Hangup
== Manager registered action Status
== Manager registered action Setvar
== Manager registered action Getvar
== Manager registered action Redirect
== Manager registered action Originate
== Manager registered action Command
== Manager registered action ExtensionState
== Manager registered action AbsoluteTimeout
== Manager registered action MailboxStatus
== Manager registered action MailboxCount
== Manager registered action ListCommands
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/cdr.conf': Found
Jul 12 10:04:02 [COLOR="Yellow"]NOTICE[13791]:[/COLOR] cdr.c:1191 do_reload: CDR simple logging enabled.
== Parsing '/etc/asterisk/rtp.conf': Found
== RTP Allocating from port range 10000 -> 20000
Asterisk PBX Core Initializing
Registering builtin applications:
[AbsoluteTimeout]
== Registered application 'AbsoluteTimeout'
[Answer]
== Registered application 'Answer'
[BackGround]
== Registered application 'BackGround'
[Busy]
== Registered application 'Busy'
[Congestion]
== Registered application 'Congestion'
[DigitTimeout]
== Registered application 'DigitTimeout'
[Goto]
== Registered application 'Goto'
[GotoIf]
== Registered application 'GotoIf'
[GotoIfTime]
== Registered application 'GotoIfTime'
[ExecIfTime]
== Registered application 'ExecIfTime'
[Hangup]
== Registered application 'Hangup'
[NoOp]
== Registered application 'NoOp'
[Progress]
== Registered application 'Progress'
[ResetCDR]
== Registered application 'ResetCDR'
[ResponseTimeout]
== Registered application 'ResponseTimeout'
[Ringing]
== Registered application 'Ringing'
[SayNumber]
== Registered application 'SayNumber'
[SayDigits]
== Registered application 'SayDigits'
[SayAlpha]
== Registered application 'SayAlpha'
[SayPhonetic]
== Registered application 'SayPhonetic'
[SetAccount]
== Registered application 'SetAccount'
[SetAMAFlags]
== Registered application 'SetAMAFlags'
[SetGlobalVar]
== Registered application 'SetGlobalVar'
[SetLanguage]
== Registered application 'SetLanguage'
[Set]
== Registered application 'Set'
[SetVar]
== Registered application 'SetVar'
[ImportVar]
== Registered application 'ImportVar'
[Wait]
== Registered application 'Wait'
[WaitExten]
== Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[chan_capi.so] => (Common ISDN API for Asterisk)
== Parsing '/etc/asterisk/capi.conf': Found
-- Registering at CAPI (blocksize=160)
== This box has 1 capi controller(s).
-- CAPI/contr1 supports DTMF
-- CAPI/contr1 supports supplementary services
> supplementary services : 0x000003ff
> HOLD/RETRIEVE TERMINAL-PORTABILITY ECT 3PTY CF CD MCID CCBS MWI CCNR
-- CAPI/contr1 private options=0x01000001
== Reading config for ISDN1
-- capi_pvt ISDN1-pseudo-D (*,default,2,2) (0,4,0)
-- capi_pvt ISDN1 (*,default,2,2) (0,4,0)
-- capi_pvt ISDN1 (*,default,2,2) (0,4,0)
-- listening on contr1 CIPmask = 0x1fff03ff
== Registered channel type 'CAPI' (Common ISDN API Driver ($Revision: 296 $))
== Registered application 'capiCommand'
[res_musiconhold.so] => (Music On Hold Resource)
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
== Parsing '/etc/asterisk/musiconhold.conf': Found
[res_adsi.so] => (ADSI Resource)
== Parsing '/etc/asterisk/adsi.conf': Found
[res_agi.so] => (Asterisk Gateway Interface (AGI))
== Registered application 'DeadAGI'
== Registered application 'EAGI'
== Registered application 'XAGI'
== Registered application 'AGI'
[res_crypto.so] => (Cryptographic Digital Signatures)
-- Loaded PUBLIC key 'freeworlddialup'
-- Loaded PUBLIC key 'iaxtel'
[res_features.so] => (Call Features Resource)
== Parsing '/etc/asterisk/features.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '700' priority 1 to parkedcalls
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action ParkedCalls
== Registered application 'HoldedCall'
== Registered application 'AutoanswerLogin'
== Registered application 'Autoanswer'
[res_indications.so] => (Indications Configuration)
== Parsing '/etc/asterisk/indications.conf': Found
-- Registered indication country 'at'
-- Registered indication country 'au'
-- Registered indication country 'br'
-- Registered indication country 'be'
-- Registered indication country 'ch'
-- Registered indication country 'cl'
-- Registered indication country 'cn'
-- Registered indication country 'cz'
-- Registered indication country 'de'
-- Registered indication country 'dk'
-- Registered indication country 'ee'
-- Registered indication country 'es'
-- Registered indication country 'fi'
-- Registered indication country 'fr'
-- Registered indication country 'gr'
-- Registered indication country 'hu'
-- Registered indication country 'it'
-- Registered indication country 'lt'
-- Registered indication country 'mx'
-- Registered indication country 'nl'
-- Registered indication country 'no'
-- Registered indication country 'nz'
-- Registered indication country 'pl'
-- Registered indication country 'pt'
-- Registered indication country 'ru'
-- Registered indication country 'se'
-- Registered indication country 'sg'
-- Registered indication country 'uk'
-- Registered indication country 'us'
-- Registered indication country 'us-o'
-- Registered indication country 'tw'
-- Registered indication country 'za'
-- Setting default indication country to 'us'
== Registered application 'PlayTones'
== Registered application 'StopPlayTones'
[res_monitor.so] => (Call Monitoring Resource)
== Registered application 'Monitor'
== Registered application 'StopMonitor'
== Registered application 'ChangeMonitor'
== Manager registered action Monitor
== Manager registered action StopMonitor
== Manager registered action ChangeMonitor
[res_watchdog.so] => (Watchdog Resource)
[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_config.so] => (Text Extension Configuration)
== Parsing '/etc/asterisk/extensions.conf': Found
-- Registered extension context 'internal'
-- Added extension '742606' priority 1 to internal
-- Added extension '742606' priority 2 to internal
-- Added extension '742606' priority 3 to internal
-- Added extension '742606' priority 102 to internal
-- Added extension '742606' priority 103 to internal
-- Added extension '742604' priority 1 to internal
-- Added extension '742604' priority 2 to internal
-- Added extension '742604' priority 3 to internal
-- Added extension '742604' priority 102 to internal
-- Added extension '742604' priority 103 to internal
-- Added extension '630325' priority 1 to internal
-- Added extension '630325' priority 2 to internal
-- Added extension '630325' priority 3 to internal
-- Added extension '630325' priority 102 to internal
-- Added extension '630325' priority 103 to internal
-- Added extension '630324' priority 1 to internal
-- Added extension '630324' priority 2 to internal
-- Added extension '630324' priority 3 to internal
-- Added extension '630324' priority 102 to internal
-- Added extension '630324' priority 103 to internal
-- Registered extension context 'external-ISDN'
-- Added extension '_0.' priority 1 to external-ISDN
-- Added extension '_0.' priority 2 to external-ISDN
-- Added extension '_0.' priority 3 to external-ISDN
-- Registered extension context 'external-SIP'
-- Added extension '_8.' priority 1 to external-SIP
-- Added extension '_8.' priority 2 to external-SIP
-- Added extension '_9.' priority 1 to external-SIP
-- Added extension '_9.' priority 2 to external-SIP
-- Registered extension context 'vmailbox'
-- Added extension '8000' priority 1 to vmailbox
-- Registered extension context 'default'
-- Including context 'vmailbox' in context 'default'
-- Including context 'internal' in context 'default'
-- Including context 'external-ISDN' in context 'default'
-- Including context 'external-SIP' in context 'default'
[skipping pbx_dundi.so]
[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
== Registered custom function LANGUAGE
== Registered custom function MUSICCLASS
[pbx_loopback.so] => (Loopback Switch)
[pbx_realtime.so] => (Realtime Switch)
[pbx_spool.so] => (Outgoing Spool Support)
Junk at the beginning 49443302
Warning, flexibel rate not heavily tested!
[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
Jul 12 10:04:02 [COLOR="Red"]WARNING[13791]:[/COLOR] db.c:67 dbinit: Unable to open Asterisk database
Jul 12 10:04:02 [COLOR="Red"]WARNING[13791]:[/COLOR] db.c:423 ast_db_gettree: Database unavailable
[chan_features.so] => (Feature Proxy Channel)
== Registered channel type 'Feature' (Feature Proxy Channel Driver)
[chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
== Registered custom function IAXPEER
== Registered application 'IAX2Provision'
== Manager registered action IAXpeers
== Manager registered action IAXnetstats
== Parsing '/etc/asterisk/iax.conf': Found
== Using TOS bits 16
== Binding IAX2 to default address 0.0.0.0:4569
-- doing lookup for '216.207.245.47'
== Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
== IAX Ready and Listening
== Loaded firmware 'iaxy.bin'
== Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
Jul 12 10:04:02 [COLOR="Red"]WARNING[13791]:[/COLOR] db.c:67 dbinit: Unable to open Asterisk database
[chan_local.so] => (Local Proxy Channel)
== Registered channel type 'Local' (Local Proxy Channel Driver)
[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))
[skipping chan_oss.so]
[chan_phone.so] => (Linux Telephony API Support)
== Parsing '/etc/asterisk/phone.conf': Found
== Registered channel type 'Phone' (Standard Linux Telephony API Driver)
[skipping chan_sccp.so]
[chan_sip.so] => (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
Jul 12 10:04:02 WARNING[13791]: chan_sip.c:12630 reload_config: Invalid address for externhost keyword:
== SIP Listening on 0.0.0.0:5060
== Using TOS bits 24
== 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
[skipping chan_skinny.so]
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, PRI Signalling signalling
-- Registered channel 2, PRI Signalling signalling
-- Automatically generated pseudo channel
== Starting D-Channel on span 1
== 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'
[app_adsiprog.so] => (Asterisk ADSI Programming Application)
== Registered application 'ADSIProg'
[app_alarmreceiver.so] => (Alarm Receiver for Asterisk)
== Parsing '/etc/asterisk/alarmreceiver.conf': Found
== Registered application 'AlarmReceiver'
[app_authenticate.so] => (Authentication Application)
== Registered application 'Authenticate'
[app_callingpres.so] => (Callingpres(number): Changes the presentation for the callerid. Should )
== Registered application 'CallingPres'
[app_cdr.so] => (Tell Asterisk to not maintain a CDR for the current call)
== Registered application 'NoCDR'
[app_chanisavail.so] => (Check channel availability)
== Registered application 'ChanIsAvail'
[app_chanspy.so] => (Listen to the audio of an active channel
)
== Registered application 'ChanSpy'
[app_controlplayback.so] => (Control Playback Application)
== Registered application 'ControlPlayback'
[app_cut.so] => (Cut out information from a string)
== Registered custom function CUT
== Registered custom function SORT
== Registered application 'Sort'
== Registered application 'Cut'
[app_db.so] => (Database Access Functions)
== Registered application 'DBget'
== Registered application 'DBput'
== Registered application 'DBdel'
== Registered application 'DBdeltree'
[app_devstate.so] => (Application for sending device state messages)
== Registered channel type 'DS' (Application for sending device state messages)
== Manager registered action Devstate
== Registered application 'Devstate'
[app_dial.so] => (Dialing Application)
== Registered application 'Dial'
== Registered application 'RetryDial'
[app_dictate.so] => (Virtual Dictation Machine)
== Registered application 'Dictate'
[app_directed_pickup.so] => (Directed Call Pickup Application)
== Registered application 'DPickup'
[app_directory.so] => (Extension Directory)
== Registered application 'Directory'
[app_disa.so] => (DISA (Direct Inward System Access) Application)
== Registered application 'DISA'
[app_dumpchan.so] => (Dump Info About The Calling Channel)
== Registered application 'DumpChan'
[app_echo.so] => (Simple Echo Application)
== Registered application 'Echo'
[app_enumlookup.so] => (ENUM Lookup)
== Registered application 'EnumLookup'
== Parsing '/etc/asterisk/enum.conf': Found
[app_eval.so] => (Reevaluates strings)
== Registered application 'Eval'
[app_exec.so] => (Executes applications)
== Registered application 'Exec'
[app_externalivr.so] => (External IVR Interface Application)
== Registered application 'ExternalIVR'
[app_festival.so] => (Simple Festival Interface)
== Registered application 'Festival'
[app_flash.so] => (Flash zap trunk application)
== Registered application 'Flash'
[app_forkcdr.so] => (Fork The CDR into 2 separate entities.)
== Registered application 'ForkCDR'
[app_getcpeid.so] => (Get ADSI CPE ID)
== Registered application 'GetCPEID'
[app_groupcount.so] => (Group Management Routines)
== Registered application 'GetGroupCount'
== Registered application 'SetGroup'
== Registered application 'CheckGroup'
== Registered application 'GetGroupMatchCount'
[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'
[app_ices.so] => (Encode and Stream via icecast and ices)
== Registered application 'ICES'
[app_image.so] => (Image Transmission Application)
== Registered application 'SendImage'
[app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database)
== Registered application 'LookupBlacklist'
[app_lookupcidname.so] => (Look up CallerID Name from local database)
== Registered application 'LookupCIDName'
[app_macro.so] => (Extension Macros)
== Registered application 'MacroExit'
== Registered application 'MacroIf'
== Registered application 'Macro'
[app_math.so] => (Basic Math Functions)
== Registered application 'Math'
[app_md5.so] => (MD5 checksum applications)
== Registered application 'MD5Check'
== Registered application 'MD5'
[app_meetme.so] => (MeetMe conference bridge)
== Parsing '/etc/asterisk/meetme.conf': Found
== Registered application 'MeetMeAdmin'
== Registered application 'MeetMeCount'
== Registered application 'MeetMe'
[app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application)
== Registered application 'Milliwatt'
[app_mixmonitor.so] => (Mixed Audio Monitoring Application)
== Registered application 'MixMonitor'
[app_mp3.so] => (Silly MP3 Application)
== Registered application 'MP3Player'
[app_nbscat.so] => (Silly NBS Stream Application)
== Registered application 'NBScat'
[app_page.so] => (Page Multiple Phones)
== Registered application 'Page'
[app_parkandannounce.so] => (Call Parking and Announce Application)
== Registered application 'ParkAndAnnounce'
[app_pickup.so] => (PickUp/PickDown/Steal/PickupChan/StealChan)
== Registered application 'StealChan'
== Registered application 'PickupChan'
== Registered application 'PickDown'
== Registered application 'Steal'
== Registered application 'PickUp'
[app_playback.so] => (Sound File Playback Application)
== Registered application 'Playback'
[app_privacy.so] => (Require phone number to be entered, if no CallerID sent)
== Registered application 'PrivacyManager'
[app_queue.so] => (True Call Queueing)
== Registered application 'Queue'
== Manager registered action Queues
== Manager registered action QueueStatus
== Manager registered action QueueAdd
== Manager registered action QueueRemove
== Manager registered action QueuePause
== Registered application 'AddQueueMember'
== Registered application 'RemoveQueueMember'
== Registered application 'PauseQueueMember'
== Registered application 'UnpauseQueueMember'
== Registered custom function QUEUEAGENTCOUNT
== Parsing '/etc/asterisk/queues.conf': Found
Jul 12 10:04:02 [COLOR="Red"]WARNING[13791]:[/COLOR] db.c:67 dbinit: Unable to open Asterisk database
Jul 12 10:04:02 [COLOR="Red"]WARNING[13791]:[/COLOR] db.c:423 ast_db_gettree: Database unavailable
[app_random.so] => (Random goto)
== Registered application 'Random'
[app_read.so] => (Read Variable Application)
== Registered application 'Read'
[app_readfile.so] => (Stores content of file into a variable)
== Registered application 'ReadFile'
[app_realtime.so] => (Realtime Data Lookup/Rewrite)
== Registered application 'RealTimeUpdate'
== Registered application 'RealTime'
[app_record.so] => (Trivial Record Application)
== Registered application 'Record'
[app_sayunixtime.so] => (Say time)
== Registered application 'SayUnixTime'
== Registered application 'DateTime'
[app_segfault.so] => (Application for crashing Asterisk with a segmentation fault)
== Registered application 'Segfault'
[app_senddtmf.so] => (Send DTMF digits Application)
== Registered application 'SendDTMF'
[app_sendtext.so] => (Send Text Applications)
== Registered application 'SendText'
[app_setcallerid.so] => (Set CallerID Application)
== Registered application 'SetCallerPres'
== Registered application 'SetCallerID'
[app_setcdruserfield.so] => (CDR user field apps)
== Registered application 'SetCDRUserField'
== Registered application 'AppendCDRUserField'
== Manager registered action SetCDRUserField
[app_setcidname.so] => (Set CallerID Name)
== Registered application 'SetCIDName'
[app_setcidnum.so] => (Set CallerID Number)
== Registered application 'SetCIDNum'
[app_setrdnis.so] => (Set RDNIS Number)
== Registered application 'SetRDNIS'
[app_settransfercapability.so] => (Set ISDN Transfer Capability)
== Registered application 'SetTransferCapability'
[app_sms.so] => (SMS/PSTN handler)
== Registered application 'SMS'
[app_softhangup.so] => (Hangs up the requested channel)
== Registered application 'SoftHangup'
[app_stack.so] => (Stack Routines)
== Registered application 'StackPop'
== Registered application 'Return'
== Registered application 'GosubIf'
== Registered application 'Gosub'
[app_system.so] => (Generic System() application)
== Registered application 'TrySystem'
== Registered application 'System'
[app_talkdetect.so] => (Playback with Talk Detection)
== Registered application 'BackgroundDetect'
[app_test.so] => (Interface Test Application)
== Registered application 'TestClient'
== Registered application 'TestServer'
[app_transfer.so] => (Transfer)
== Registered application 'Transfer'
[app_txtcidname.so] => (TXTCIDName)
== Registered application 'TXTCIDName'
[app_url.so] => (Send URL Applications)
== Registered application 'SendURL'
[app_userevent.so] => (Custom User Event Application)
== Registered application 'UserEvent'
[app_verbose.so] => (Send verbose output)
== Registered application 'Verbose'
[app_voicemail.so] => (Comedian Mail (Voicemail System))
== Registered application 'VoiceMail'
== Registered application 'VoiceMailMain'
== Registered application 'MailboxExists'
== Registered application 'VMAuthenticate'
== Parsing '/etc/asterisk/voicemail.conf': Found
[app_waitforring.so] => (Waits until first ring after time)
== Registered application 'WaitForRing'
[app_waitforsilence.so] => (Wait For Silence)
== Registered application 'WaitForSilence'
[app_while.so] => (While Loops and Conditional Execution)
== Registered application 'While'
== Registered application 'ExecIf'
== Registered application 'EndWhile'
[app_zapateller.so] => (Block Telemarketers with Special Information Tone)
== Registered application 'Zapateller'
[app_zapbarge.so] => (Barge in on Zap channel application)
== Registered application 'ZapBarge'
[app_zapras.so] => (Zap RAS Application)
== Registered application 'ZapRAS'
[app_zapscan.so] => (Scan Zap channels application)
== Registered application 'ZapScan'
[cdr_csv.so] => (Comma Separated Values CDR Backend)
[cdr_custom.so] => (Customizable Comma Separated Values CDR Backend)
== Parsing '/etc/asterisk/cdr_custom.conf': Found
[cdr_manager.so] == Primary D-Channel on span 1 up for TEI 64
=> (Asterisk Call Manager CDR Backend)
== Parsing '/etc/asterisk/cdr_manager.conf': Found
[skipping chan_oss.so]
[skipping chan_sccp.so]
[skipping chan_skinny.so]
[codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder)
== Parsing '/etc/asterisk/codecs.conf': Found
-- codec_adpcm: using generic PLC
== Registered translator 'adpcmtolin' from format adpcm to slin, cost 1
== Registered translator 'lintoadpcm' from format slin to adpcm, cost 1
[codec_alaw.so] => (A-law Coder/Decoder)
== Parsing '/etc/asterisk/codecs.conf': Found
-- codec_alaw: using generic PLC
== Registered translator 'alawtolin' from format alaw to slin, cost 1
== Registered translator 'lintoalaw' from format slin to alaw, cost 1
[codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder)
== Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1
== Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1
[codec_g726.so] => (ITU G.726-32kbps G726 Transcoder)
== Parsing '/etc/asterisk/codecs.conf': Found
-- codec_g726: using generic PLC
== Registered translator 'g726tolin' from format g726 to slin, cost 6
== Registered translator 'lintog726' from format slin to g726, cost 6
[codec_g729a.so] => (Annex A/B (floating point) G.729 Codec (optimized for i586))
Jul 12 10:04:02 [COLOR="Yellow"]NOTICE[13791][/COLOR]: codec_g729.c:460 load_module: G.729 transcoding module Copyright (C) 1999-2005 Digium, Inc.
Jul 12 10:04:02 [COLOR="Yellow"]NOTICE[13791]:[/COLOR] codec_g729.c:461 load_module: This module is supplied under a commercial license granted by Digium, Inc.
Jul 12 10:04:02 [COLOR="Yellow"]NOTICE[13791]:[/COLOR] codec_g729.c:462 load_module: Please see the full license text supplied by the accompanying
Jul 12 10:04:02 [COLOR="Yellow"]NOTICE[13791]:[/COLOR] codec_g729.c:463 load_module: "register" utility, or ask for a copy from Digium.
== G.729 Host-ID: be:e2:01:73:9e:19:1f:ac:d7:3a:99
== Found total of 0 G.729 licenses
[codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator)
== Parsing '/etc/asterisk/codecs.conf': Found
-- codec_gsm: using generic PLC
== Registered translator 'gsmtolin' from format gsm to slin, cost 2
== Registered translator 'lintogsm' from format slin to gsm, cost 8
[codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator)
== Registered translator 'ilbctolin' from format ilbc to slin, cost 11
== Registered translator 'lintoilbc' from format slin to ilbc, cost 71
[codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder)
== Parsing '/etc/asterisk/codecs.conf': Found
-- codec_lpc10: using generic PLC
== Registered translator 'lpc10tolin' from format lpc10 to slin, cost 6
== Registered translator 'lintolpc10' from format slin to lpc10, cost 10
[codec_speex.so] => (Speex/PCM16 (signed linear) Codec Translator)
== Parsing '/etc/asterisk/codecs.conf': Found
-- CODEC SPEEX: Setting Quality to 3
-- CODEC SPEEX: Setting Complexity to 2
-- CODEC SPEEX: Perceptual Enhancement Mode. [on]
-- CODEC SPEEX: VAD Mode. [on]
-- CODEC SPEEX: VBR Mode. [on]
-- CODEC SPEEX: Disabling ABR
-- CODEC SPEEX: Setting VBR Quality to 4.000000
-- CODEC SPEEX: DTX Mode. [off]
-- CODEC SPEEX: Preprocessing. [off]
-- CODEC SPEEX: Preprocessor VAD. [off]
-- CODEC SPEEX: Preprocessor AGC. [off]
-- CODEC SPEEX: Setting preprocessor AGC Level to 8000.000000
-- CODEC SPEEX: Preprocessor Denoise. [off]
-- CODEC SPEEX: Preprocessor Dereverb. [off]
-- CODEC SPEEX: Setting preprocessor Dereverb Decay to 0.400000
-- CODEC SPEEX: Setting preprocessor Dereverb Level to 0.300000
== Registered translator 'speextolin' from format speex to slin, cost 9
== Registered translator 'lintospeex' from format slin to speex, cost 130
[codec_ulaw.so] => (Mu-law Coder/Decoder)
== Parsing '/etc/asterisk/codecs.conf': Found
-- codec_ulaw: using generic PLC
== Registered translator 'ulawtolin' from format ulaw to slin, cost 1
== Registered translator 'lintoulaw' from format slin to ulaw, cost 1
[format_au.so] => (Sun Microsystems AU format (signed linear))
== Registered file format au, extension(s) au
[format_g723.so] => (G.723.1 Simple Timestamp File Format)
== Registered file format g723sf, extension(s) g723|g723sf
[format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
== Registered file format g726-40, extension(s) g726-40
== Registered file format g726-32, extension(s) g726-32
== Registered file format g726-24, extension(s) g726-24
== Registered file format g726-16, extension(s) g726-16
[format_g729.so] => (Raw G729 data)
== Registered file format g729, extension(s) g729
[format_gsm.so] => (Raw GSM data)
== Registered file format gsm, extension(s) gsm
[format_h263.so] => (Raw h263 data)
== Registered file format h263, extension(s) h263
[format_ilbc.so] => (Raw iLBC data)
== Registered file format iLBC, extension(s) ilbc
[format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format)
== Registered format 'jpg' (JPEG (Joint Picture Experts Group))
[format_pcm.so] => (Raw uLaw 8khz Audio support (PCM))
== Registered file format pcm, extension(s) pcm|ulaw|ul|mu
[format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support)
== Registered file format alaw, extension(s) alaw|al
[format_sln.so] => (Raw Signed Linear Audio support (SLN))
== Registered file format sln, extension(s) sln|raw
[format_vox.so] => (Dialogic VOX (ADPCM) File Format)
== Registered file format vox, extension(s) vox
[format_wav.so] => (Microsoft WAV format (8000hz Signed Linear))
== Registered file format wav, extension(s) wav
[format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM))
== Registered file format wav49, extension(s) WAV|wav49
[func_callerid.so] => (Caller ID related dialplan function)
== Registered custom function CALLERID
[func_enum.so] => (ENUM Related Functions)
== Registered custom function ENUMLOOKUP
== Registered custom function TXTCIDNAME
[func_uri.so] => (URI encode/decode functions)
== Registered custom function URIDECODE
== Registered custom function URIENCODE
[skipping pbx_dundi.so]
Jul 12 10:04:03 [COLOR="Red"]WARNING[13791]:[/COLOR] db.c:67 dbinit: Unable to open Asterisk database
== Manager registered action DBget
== Manager registered action DBput
== Manager registered action DBdel
== Parsing '/etc/asterisk/enum.conf': Found
Asterisk Ready.
Jul 12 10:14:06 NOTICE[13807]: chan_sip.c:9798 handle_response_peerpoke: Peer 'sipgate-mohar' is now TOO LAGGED! (3102ms / 2000ms)
Jul 12 10:14:06 NOTICE[13807]: chan_sip.c:9798 handle_response_peerpoke: Peer 'sipgate-raisch' is now TOO LAGGED! (3104ms / 2000ms)
Jul 12 10:14:16 NOTICE[13807]: chan_sip.c:9792 handle_response_peerpoke: Peer 'sipgate-mohar' is now REACHABLE! (91ms / 2000ms)
Jul 12 10:14:16 NOTICE[13807]: chan_sip.c:9792 handle_response_peerpoke: Peer 'sipgate-raisch' is now REACHABLE! (93ms / 2000ms)
Anzeige von Astersisk bei einem ISDN-Anruf:
== ISDN1: Incoming call '01762819xxxx -> '742xxx'
-- ISDN1: Updated channel name: CAPI/ISDN1/742xxx-1
-- Executing Dial("CAPI/ISDN1/742606-1", "Zap/g1/742xxx|20|Tt") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/742xxx
-- Zap/1-1 is proceeding passing it to CAPI/ISDN1/742xxx-1
-- Zap/1-1 is ringing
Anzeige von Asterisk bei einem Anruf über SIP:
nichts
Anzeige von Asterisk bei einem Anruf über SIP (mit "context=ankommend"):
*CLI> Jul 12 10:51:24 NOTICE[14017]: pbx.c:1739 pbx_extension_helper: Cannot find extension context 'ankommend'
Anzeige von Asterisk wenn ich telefoniere (über SIP):
*CLI> -- Accepting overlap voice call from '' to '90' on channel 0/2, span 1
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial("Zap/2-1", "SIP/01762819xxxx@sipgate-raisch||Ttr") in new stack
-- Called 01762819xxxx@sipgate-raisch
-- SIP/sipgate-raisch-2444 answered Zap/2-1
-- Channel 0/2, span 1 got hangup request
== Spawn extension (default, 901762819xxxx, 1) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
2. Frage:
Momentan werden die Rufnummern der ISDN-Anrufer an den Telefonen angezeigt; allerdings MIT der "Amtkennziffer". Also wird aus "06073blablubb" ein "006073blablubb"
Ist das irgendwo einstellbar, das die erste Ziffer nicht angezeigt wird?
Vielen Dank schon mal...
ciao
Indy