Asterisk @ Speedport W701V => eingehende ISDN-Gespräche laufen nicht

fr34k4d3113

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Hi,

Ich habe hier einen gefreetzten Speedport W701V mit Asterisk 1.4, er hängt an einer Auerswald 2206 am internen S0. Ausgehende Gespräche klappen problemlos, leider keine eingehenden.
Auf dem CLI bekomme ich nur folgende Meldungen bei einem Anruf:
Code:
  == ISDN1#02: Incoming call '015112341234' -> '104'
[08497331]DSP: XDU=1( S1 ) OVR=0 MIPS_OVR=0
[08498331]DSP: XDU=3( S1 S2 ) OVR=0 MIPS_OVR=0
  == ISDN1#02: CAPI Hangingup for PLCI=0xd01 in state 4
Es scheint so als würde er mitbekommen das ich auf der MSN 104 (in der TK-Anlage eingerichtet) anrufe, der Anruf wird aber nicht auf dem in der extensions.conf zugeordneten SIP-Account signalisiert.

Auszug der capi.conf:
Code:
[ISDN1]         
ntmode=no     
isdnmode=msn    
incomingmsn=104   
defaultcid=104 
controller=1    
group=1         
softdtmf=off    
relaxdtmf=off 
accountcode= 
context=capi_in1 
;holdtype=hold   
bridge=no  
devices=2

Auszug extensions.conf:
Code:
[capi_in1]
exten => _X.,80,Dial(SIP/7701,30,r)

An der SIP.conf wurde noch nichts geändert, die ist noch im "Auslieferungszustand"
 
Wird auf der 701v das S0-Interace via Controller 1 angesprochen oder warum setzt Du controller=1 ? Normalerweise wird das S0-Interface via controller=3 angesprochen!

Gruß
dynamic
 
Ich habe einfach die defaultconfig übernommen. Wenn ich nach der gehe müsste der externe S0 der Fritzbox/Speedport controller=1 sein und der interne S0 (den der Speedport ja nicht hat) controller=3. Habe es auch mal ausprobiert, aber da geht gar nix, da wird das eingehende Gespräch gar nicht erkannt.
 
Habe mich wohl von der Aussage
er hängt an einer Auerswald 2206 am internen S0
verleiten lassen :). Du hast also am externen S0 der 701v den internen S0 Bus Deiner Auerswald angeschlossen ?
Wie verhält sich der Asterisk, wenn Du "incomingmsn=*" setzt ?

Gruß
dynamic
 
Habe ich schon probiert, kommt das gleiche raus. Auf dem CLI sehe ich, dass er einen Anruf der MSN 104 erhält, aber signalisiert wird der Anruf auch nicht.
 
Gib am CLI-Prompt folgendes ein:
Code:
core set verbose 10
dialplan show capi_in1
und Poste den Output inkl. eines kompletten CLI Log von einem eingehenden Anruf.

Gruß
dynamic
 
Code:
[ Context 'capi_in1' created by 'pbx_config' ]
  '_X.' =>          80. Dial(SIP/7701|30|r)                       [pbx_config]

-= 1 extension (1 priority) in 1 context. =-

CLI-Output von einem Anruf:
Code:
  == ISDN1#02: Incoming call '09xxxxxxx' -> '104'
[15057778]DSP: XDU=1( S1 ) OVR=0 MIPS_OVR=0
[15058778]DSP: XDU=5( S1 S2 ) OVR=0 MIPS_OVR=0
  == ISDN1#02: CAPI Hangingup for PLCI=0x1601 in state 4 [I] // ich habe hier allerdings nicht aufgelegt, es klingelt auch weiter[/I]
[15059778]DSP: XDU=6( S1 S2 ) OVR=0 MIPS_OVR=0
[15060778]DSP: XDU=4( S1 S2 ) OVR=0 MIPS_OVR=0
[I]// jetzt habe ich aufgelegt[/I]
 
Ändere die exten wie folgt und versuche es dann nochmal:
Code:
[capi_in1]
exten => s,1,Dial(SIP/7701,30,r)
Gruß
dynamic
 
Bringt leider auch keine Veränderung. Irgendwo im Forum habe ich auch den Tipp gelesen in der capi.conf "immediate=yes" zu setzen, leider auch ohne Erfolg.
Code:
  == ISDN1#02: Incoming call '015xxxxxxxxx' -> '104'
[15340350]DSP: XDU=1( S1 ) OVR=0 MIPS_OVR=0
[15341350]DSP: XDU=3( S1 S2 ) OVR=0 MIPS_OVR=0
  == ISDN1#02: CAPI Hangingup for PLCI=0x601 in state 4
[15342350]DSP: XDU=4( S1 S2 ) OVR=0 MIPS_OVR=0
 
Es ist schwierig so ins Blaue hinein zu erraten, was evtl. das Problem sein könnte ...
Ich gehe davon aus, dass in Deiner Konfiguration ein Fehler ist.

Wie sieht der Output beim Starten des Asterisk ( export TERM=xterm;asterisk -cvvvvv ) aus, gibt es dort irgendwelche Auffälligkeiten ?
Poste evtl. nochmal Deine gesamte extensions.conf

Eventuell kann man da was sehen.

btw.
Code:
[15340350]DSP: XDU=1( S1 ) OVR=0 MIPS_OVR=0
[15341350]DSP: XDU=3( S1 S2 ) OVR=0 MIPS_OVR=0
[15342350]DSP: XDU=4( S1 S2 ) OVR=0 MIPS_OVR=0
sind keine Asterisk Meldungen !

Gruß
dynamic
 
Hi,
Ich poste mal die Ausgaben vom asterisk und die extensions.conf, evtl. fällt dir ja nochwas auf.

asterisk -cvvvvvv:
Code:
Asterisk 1.4.22, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
  == 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
  == Manager registered action Park
res_features.so => (Call Features Resource)
  == 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_alaw.so => (A-law Coder/Decoder)
  == Parsing '/etc/asterisk/dnsmgr.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 GetConfig
  == Manager registered action UpdateConfig
  == 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
  == Manager registered action UserEvent
  == Manager registered action WaitEvent
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/cdr.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 9078 -> 9097
  == UDPTL allocating from port range 4500 -> 4999
Asterisk PBX Core Initializing
Registering builtin applications:
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
 == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [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'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SayAlpha]
  == Registered application 'SayAlpha'
 [SayPhonetic]
  == Registered application 'SayPhonetic'
 [SetAMAFlags]
  == Registered application 'SetAMAFlags'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [Set]
  == Registered application 'Set'
 [ImportVar]
  == Registered application 'ImportVar'
 [Wait]
  == Registered application 'Wait'
 [WaitExten]
  == Registered application 'WaitExten'
  == Manager registered action DBGet
  == Manager registered action DBPut
  == Parsing '/etc/asterisk/enum.conf': Found
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
  == Registered application 'StartMusicOnHold'
  == Registered application 'StopMusicOnHold'
  == Parsing '/etc/asterisk/musiconhold.conf': Found
res_musiconhold.so => (Music On Hold Resource)
  == Parsing '/etc/asterisk/indications.conf': Found
    -- Registered indication country 'at'
    -- Registered indication country 'au'
    -- Registered indication country 'bg'
    -- 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 'il'
    -- Registered indication country 'in'
    -- Registered indication country 'it'
    -- Registered indication country 'lt'
    -- Registered indication country 'jp'
    -- Registered indication country 'mx'
    -- Registered indication country 'my'
    -- Registered indication country 'nl'
    -- Registered indication country 'no'
    -- Registered indication country 'nz'
    -- Registered indication country 'ph'
    -- Registered indication country 'pl'
    -- Registered indication country 'pt'
    -- Registered indication country 'ru'
    -- Registered indication country 'se'
    -- Registered indication country 'sg'
    -- Registered indication country 'th'
    -- Registered indication country 'uk'
    -- Registered indication country 'us'
    -- Registered indication country 'us-o'
    -- Registered indication country 'tw'
    -- Registered indication country 've'
    -- Registered indication country 'za'
    -- Setting default indication country to 'us'
  == Registered application 'PlayTones'
  == Registered application 'StopPlayTones'
res_indications.so => (Indications Resource)
pbx_spool.so => (Outgoing Spool Support)
  == Parsing '/etc/asterisk/extensions.conf': Found
  == Setting global variable 'CAPI_CALLERID' to '104'
  == Setting global variable 'CB_IN_CHANNEL' to 'SIP/7702'
  == Setting global variable 'CB_OUT_CHANNEL' to 'SIP/7702'
  == Setting global variable 'CB_CALLERID' to 'Fritzbox <1234567>'
  == Setting global variable 'CB_WAITTIME' to '90'
  == Setting global variable 'CB_CONTEXT' to 'echotest'
  == Setting global variable 'CB_EXTENSION' to '80'
  == Setting global variable 'MAILER_TO' to 'root@localhost'
  == Setting global variable 'MAILER_FROM' to 'asterisk@localhost'
  == Setting global variable 'MAILER_SMTP' to 'smtp.localhost'
  == Setting global variable 'MAILER_USER' to ''
  == Setting global variable 'MAILER_PASSWORD' to ''
    -- Registered extension context 'macro-entfernegitter'
    -- Added extension 's' priority 1 to macro-entfernegitter
    -- Registered extension context 'headsettest'
    -- Added extension '_95!' priority 1 to headsettest
    -- Registered extension context 'callbacktest'
    -- Added extension '_94!' priority 1 to callbacktest
    -- Added extension '_94!' priority 2 to callbacktest
    -- Added extension '_94!' priority 3 to callbacktest
    -- Added extension '_94!' priority 4 to callbacktest
    -- Added extension '_94!' priority 5 to callbacktest
    -- Added extension '_94!' priority 80 to callbacktest
    -- Registered extension context 'callback-wait'
    -- Added extension 's' priority 1 to callback-wait
    -- Added extension 's' priority 2 to callback-wait
    -- Added extension 's' priority 3 to callback-wait
    -- Added extension 't' priority 1 to callback-wait
    -- Added extension 't' priority 2 to callback-wait
    -- Added extension 'h' priority 1 to callback-wait
    -- Registered extension context 'callback-callfile'
    -- Added extension 's' priority 1 to callback-callfile
    -- Added extension 's' priority 2 to callback-callfile
    -- Added extension 's' priority 3 to callback-callfile
    -- Added extension 's' priority 4 to callback-callfile
    -- Added extension 's' priority 5 to callback-callfile
    -- Added extension 's' priority 6 to callback-callfile
    -- Added extension 's' priority 7 to callback-callfile
    -- Added extension 's' priority 8 to callback-callfile
    -- Added extension 's' priority 9 to callback-callfile
    -- Added extension 's' priority 80 to callback-callfile
    -- Added extension 's' priority 100 to callback-callfile
    -- Registered extension context 'callthroughtest'
    -- Added extension '_93!' priority 1 to callthroughtest
    -- Registered extension context 'callthroughauth'
    -- Added extension 's' priority 1 to callthroughauth
    -- Added extension 's' priority 2 to callthroughauth
    -- Added extension 's' priority 3 to callthroughauth
    -- Added extension 's' priority 4 to callthroughauth
    -- Added extension '_93!' priority 105 to callthroughauth
    -- Added extension '_93!' priority 106 to callthroughauth
    -- Added extension 's' priority 5 to callthroughauth
    -- Added extension 'h' priority 1 to callthroughauth
    -- Registered extension context 'callthrough'
    -- Added extension 's' priority 1 to callthrough
    -- Added extension 's' priority 2 to callthrough
    -- Added extension 's' priority 3 to callthrough
    -- Added extension 's' priority 4 to callthrough
    -- Added extension '_X' priority 1 to callthrough
    -- Added extension '_X' priority 2 to callthrough
    -- Added extension '*' priority 1 to callthrough
    -- Added extension '#' priority 1 to callthrough
    -- Added extension '#' priority 2 to callthrough
    -- Added extension '#' priority 3 to callthrough
    -- Added extension '#' priority 4 to callthrough
    -- Added extension '#' priority 5 to callthrough
    -- Added extension '#' priority 6 to callthrough
    -- Added extension '#' priority 7 to callthrough
    -- Added extension 't' priority 1 to callthrough
    -- Added extension 't' priority 2 to callthrough
    -- Added extension 't' priority 3 to callthrough
    -- Added extension 'h' priority 1 to callthrough
    -- Registered extension context 'mohtest'
    -- Added extension '_92!' priority 1 to mohtest
    -- Added extension '_92!' priority 2 to mohtest
    -- Added extension '_92!' priority 3 to mohtest
    -- Added extension '_92!' priority 4 to mohtest
    -- Added extension '_92!' priority 5 to mohtest
    -- Registered extension context 'record_message'
    -- Added extension '_91!' priority 1 to record_message
    -- Added extension '_91!' priority 2 to record_message
    -- Added extension '_91!' priority 3 to record_message
    -- Added extension '_91!' priority 4 to record_message
    -- Added extension '_91!' priority 5 to record_message
    -- Added extension '_91!' priority 6 to record_message
    -- Added extension '_91!' priority 7 to record_message
    -- Added extension '_91!' priority 8 to record_message
    -- Registered extension context 'recordtest'
    -- Added extension '_90!' priority 1 to recordtest
    -- Registered extension context 'record'
    -- Added extension 's' priority 1 to record
    -- Added extension 's' priority 2 to record
    -- Added extension 's' priority 3 to record
    -- Added extension 's' priority 4 to record
    -- Added extension 's' priority 5 to record
    -- Added extension 's' priority 6 to record
    -- Added extension 's' priority 7 to record
    -- Added extension 's' priority 8 to record
    -- Added extension 's' priority 9 to record
    -- Added extension 'h' priority 1 to record
    -- Added extension 'h' priority 20 to record
    -- Added extension 'h' priority 21 to record
    -- Added extension 'h' priority 22 to record
    -- Added extension 'h' priority 23 to record
    -- Added extension 'h' priority 40 to record
    -- Added extension 'h' priority 41 to record
    -- Added extension 'h' priority 45 to record
    -- Added extension 'h' priority 46 to record
    -- Added extension 'h' priority 47 to record
    -- Added extension 'h' priority 48 to record
    -- Added extension 'h' priority 49 to record
    -- Registered extension context 'echotest'
    -- Added extension '_80!' priority 1 to echotest
    -- Added extension '_80!' priority 2 to echotest
    -- Added extension '_80!' priority 3 to echotest
    -- Added extension '_80!' priority 4 to echotest
    -- Added extension '_80!' priority 5 to echotest
    -- Added extension '_80!' priority 6 to echotest
    -- Added extension '_80!' priority 7 to echotest
    -- Added extension '_80!' priority 8 to echotest
    -- Added extension '_80!' priority 9 to echotest
    -- Registered extension context 'authtest'
    -- Added extension '_89!' priority 1 to authtest
    -- Added extension '_89!' priority 2 to authtest
    -- Added extension '_89!' priority 3 to authtest
    -- Added extension '_89!' priority 4 to authtest
    -- Added extension '_89!' priority 5 to authtest
    -- Added extension '_89!' priority 6 to authtest
    -- Added extension '_89!' priority 7 to authtest
    -- Registered extension context 'lokal'
    -- Including context 'echotest' in context 'lokal'
    -- Including context 'authtest' in context 'lokal'
    -- Including context 'recordtest' in context 'lokal'
    -- Including context 'record_message' in context 'lokal'
    -- Including context 'mohtest' in context 'lokal'
    -- Including context 'callthroughtest' in context 'lokal'
    -- Including context 'callbacktest' in context 'lokal'
    -- Including context 'headsettest' in context 'lokal'
    -- Added extension '_6X.' priority 1 to lokal
    -- Added extension '_70X' priority 1 to lokal
    -- Added extension '_77XX' priority 1 to lokal
    -- Added extension '_8[1-9]' priority 1 to lokal
    -- Registered extension context 'festnetz_out'
    -- Added extension '_0X.' priority 1 to festnetz_out
    -- Added extension '_0X.' priority 2 to festnetz_out
    -- Registered extension context 'sip_out'
    -- Added extension '_99.' priority 1 to sip_out
    -- Registered extension context 'default'
    -- Including context 'lokal' in context 'default'
    -- Registered extension context 'sip7701'
    -- Including context 'lokal' in context 'sip7701'
    -- Including context 'festnetz_out' in context 'sip7701'
    -- Including context 'sip_out' in context 'sip7701'
    -- Registered extension context 'sip7702'
    -- Including context 'lokal' in context 'sip7702'
    -- Including context 'festnetz_out' in context 'sip7702'
    -- Including context 'sip_out' in context 'sip7702'
    -- Registered extension context 'sip7703'
    -- Including context 'lokal' in context 'sip7703'
    -- Including context 'festnetz_out' in context 'sip7703'
    -- Including context 'sip_out' in context 'sip7703'
    -- Registered extension context 'sip7704'
    -- Including context 'lokal' in context 'sip7704'
    -- Including context 'festnetz_out' in context 'sip7704'
    -- Including context 'sip_out' in context 'sip7704'
    -- Registered extension context 'sccp701'
    -- Including context 'lokal' in context 'sccp701'
    -- Including context 'festnetz_out' in context 'sccp701'
    -- Including context 'sip_out' in context 'sccp701'
    -- Registered extension context 'sccp702'
    -- Including context 'lokal' in context 'sccp702'
    -- Including context 'festnetz_out' in context 'sccp702'
    -- Including context 'sip_out' in context 'sccp702'
    -- Registered extension context 'sccp703'
    -- Including context 'lokal' in context 'sccp703'
    -- Including context 'festnetz_out' in context 'sccp703'
    -- Including context 'sip_out' in context 'sccp703'
    -- Registered extension context 'iax81'
    -- Including context 'lokal' in context 'iax81'
    -- Including context 'festnetz_out' in context 'iax81'
    -- Including context 'sip_out' in context 'iax81'
    -- Registered extension context 'iax82'
    -- Including context 'lokal' in context 'iax82'
    -- Including context 'festnetz_out' in context 'iax82'
    -- Including context 'sip_out' in context 'iax82'
    -- Registered extension context 'iax83'
    -- Including context 'lokal' in context 'iax83'
    -- Including context 'festnetz_out' in context 'iax83'
    -- Including context 'sip_out' in context 'iax83'
    -- Registered extension context 'iax84'
    -- Including context 'lokal' in context 'iax84'
    -- Including context 'festnetz_out' in context 'iax84'
    -- Including context 'sip_out' in context 'iax84'
    -- Registered extension context 'asterisk2_in'
    -- Including context 'lokal' in context 'asterisk2_in'
    -- Registered extension context 'capi_in1'
    -- Added extension 's' priority 80 to capi_in1
    -- Registered extension context 'capi_in3'
    -- Added extension '_9[1-8]!' priority 1 to capi_in3
    -- Added extension '_9[1-8]!' priority 2 to capi_in3
    -- Added extension '_90!' priority 1 to capi_in3
    -- Added extension '_90!' priority 2 to capi_in3
    -- Added extension '_99!' priority 3 to capi_in3
    -- Registered extension context 'mobile_in'
    -- Added extension 's' priority 1 to mobile_in
    -- Added extension 's' priority 2 to mobile_in
    -- Registered extension context 'sip_in'
    -- Added extension 'sip1' priority 1 to sip_in
    -- Added extension 'sip1' priority 2 to sip_in
pbx_config.so => (Text Extension Configuration)
  == Registered custom function TIMEOUT
func_timeout.so => (Channel timeout dialplan functions)
  == Registered custom function FIELDQTY
  == Registered custom function FILTER
  == Registered custom function REGEX
  == Registered custom function ARRAY
  == Registered custom function QUOTE
  == Registered custom function LEN
  == Registered custom function STRFTIME
  == Registered custom function STRPTIME
  == Registered custom function EVAL
  == Registered custom function KEYPADHASH
  == Registered custom function SPRINTF
func_strings.so => (String handling dialplan functions)
  == Registered custom function MD5
  == Registered custom function CHECK_MD5
func_md5.so => (MD5 digest dialplan functions)
  == Registered custom function MATH
func_math.so => (Mathematical dialplan function)
  == Registered custom function ISNULL
  == Registered custom function SET
  == Registered custom function EXISTS
  == Registered custom function IF
  == Registered custom function IFTIME
func_logic.so => (Logical dialplan functions)
  == Registered custom function GROUP_COUNT
  == Registered custom function GROUP_MATCH_COUNT
  == Registered custom function GROUP_LIST
  == Registered custom function GROUP
func_groupcount.so => (Channel group dialplan functions)
  == Registered custom function GLOBAL
func_global.so => (Global variable dialplan functions)
  == Registered custom function ENV
  == Registered custom function STAT
func_env.so => (Environment/filesystem dialplan functions)
  == Registered custom function ENUMLOOKUP
  == Registered custom function TXTCIDNAME
func_enum.so => (ENUM related dialplan functions)
  == Registered custom function DB
  == Registered custom function DB_EXISTS
  == Registered custom function DB_DELETE
func_db.so => (Database (astdb) related dialplan functions)
  == Registered custom function CHANNEL
func_channel.so => (Channel information dialplan function)
  == Registered custom function CDR
func_cdr.so => (CDR dialplan function)
  == Registered custom function CALLERID
func_callerid.so => (Caller ID related dialplan function)
  == Registered file format wav49, extension(s) WAV|wav49
format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM))
  == Registered file format wav, extension(s) wav
format_wav.so => (Microsoft WAV format (8000Hz Signed Linear))
  == Registered file format sln, extension(s) sln|raw
format_sln.so => (Raw Signed Linear Audio support (SLN))
  == Registered file format pcm, extension(s) pcm|ulaw|ul|mu
  == Registered file format alaw, extension(s) alaw|al
  == Registered file format au, extension(s) au
  == Registered file format g722, extension(s) g722
format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.722 16Khz)
  == Registered file format gsm, extension(s) gsm
format_gsm.so => (Raw GSM data)
  == Registered file format g729, extension(s) g729
format_g729.so => (Raw G729 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_g726.so => (Raw G.726 (16/24/32/40kbps) data)
  == 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
codec_ulaw.so => (mu-Law Coder/Decoder)
  == Registered translator 'ilbctolin' from format ilbc to slin, cost 2116
  == Registered translator 'lintoilbc' from format slin to ilbc, cost 11809
codec_ilbc.so => (iLBC Coder/Decoder)
  == Parsing '/etc/asterisk/codecs.conf': Found
    -- codec_gsm: using generic PLC
  == Registered translator 'gsmtolin' from format gsm to slin, cost 18
  == Registered translator 'lintogsm' from format slin to gsm, cost 49
codec_gsm.so => (GSM Coder/Decoder)
  == Parsing '/etc/asterisk/codecs.conf': Found
    -- codec_g726: using generic PLC
  == Registered translator 'g726tolin' from format g726 to slin, cost 32
  == Registered translator 'lintog726' from format slin to g726, cost 31
  == Registered translator 'g726aal2tolin' from format g726aal2 to slin, cost 31
  == Registered translator 'lintog726aal2' from format slin to g726aal2, cost 32
  == Registered translator 'g726aal2tog726' from format g726aal2 to g726, cost 2
  == Registered translator 'g726tog726aal2' from format g726 to g726aal2, cost 2
codec_g726.so => (ITU G.726-32kbps G726 Transcoder)
  == Parsing '/etc/asterisk/codecs.conf': Found
    -- codec_adpcm: using generic PLC
  == Registered translator 'adpcmtolin' from format adpcm to slin, cost 2
  == Registered translator 'lintoadpcm' from format slin to adpcm, cost 7
codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder)
  == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1
  == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1
codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder)
  == Parsing '/etc/asterisk/sip.conf': Found
  == SIP Listening on 0.0.0.0:5061
  == Using SIP TOS: none
  == Parsing '/etc/asterisk/sip_notify.conf': Found
  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
  == Registered application 'SIPDtmfMode'
  == Registered application 'SIPAddHeader'
  == 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
chan_sip.so => (Session Initiation Protocol (SIP))
  == Registered channel type 'Local' (Local Proxy Channel Driver)
chan_local.so => (Local Proxy Channel (Note: used internally by other modules))
  == Registered custom function IAXPEER
  == Registered application 'IAX2Provision'
  == Manager registered action IAXpeers
  == Manager registered action IAXnetstats
  == Parsing '/etc/asterisk/iax.conf': Found
       > doing dnsmgr_lookup for 'localhost'
  == Using TOS bits 0
  == Binding IAX2 to default address 0.0.0.0:4569
  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
  == 10 helper threads started
  == IAX Ready and Listening
  == Loaded firmware 'iaxy.bin'
  == Parsing '/etc/asterisk/iaxprov.conf': Found
    -- Loaded provisioning template 'default'
chan_iax2.so => (Inter Asterisk eXchange (Ver 2))
  == Parsing '/etc/asterisk/capi.conf': Found
    -- Registering at CAPI (blocksize=160 maxlogicalchannels=2)
  == This box has 5 capi controller(s).
    -- Contr1 supports DTMF
    -- Contr1 supports supplementary services
       > supplementary services : 0x000003ff
       >  HOLD/RETRIEVE TERMINAL-PORTABILITY ECT 3PTY CF CD MCID CCBS MWI CCNR
    -- Contr1 private options=0x01000001
    -- Contr2 supports DTMF
    -- Contr2 supports supplementary services
       > supplementary services : 0x000003ff
       >  HOLD/RETRIEVE TERMINAL-PORTABILITY ECT 3PTY CF CD MCID CCBS MWI CCNR
    -- Contr2 private options=0x01000001
    -- Contr3 supports DTMF
    -- Contr3 supports supplementary services
       > supplementary services : 0x000003ff
       >  HOLD/RETRIEVE TERMINAL-PORTABILITY ECT 3PTY CF CD MCID CCBS MWI CCNR
    -- Contr3 private options=0x00000001
    -- Contr4 supports DTMF
    -- Contr4 supports supplementary services
       > supplementary services : 0x000003ff
       >  HOLD/RETRIEVE TERMINAL-PORTABILITY ECT 3PTY CF CD MCID CCBS MWI CCNR
    -- Contr4 private options=0x00000001
    -- Contr5 supports DTMF
    -- Contr5 supports supplementary services
       > supplementary services : 0x000003ff
       >  HOLD/RETRIEVE TERMINAL-PORTABILITY ECT 3PTY CF CD MCID CCBS MWI CCNR
    -- Contr5 private options=0x00000001
  == Reading config for ISDN1
    -- capi D ISDN1#00 (104:capi_in1) contr=1 devs=2 EC=0,opt=4,tail=0
    -- capi B ISDN1#01 (104:capi_in1) contr=1 devs=2 EC=0,opt=4,tail=0
    -- capi B ISDN1#02 (104:capi_in1) contr=1 devs=2 EC=0,opt=4,tail=0
    -- Registering at CAPI (blocksize=160 maxlogicalchannels=3)
    -- listening on contr1 CIPmask = 0x1fff03ff
  == Registered channel type 'CAPI' (Common ISDN API Driver (1.1.1))
  == Registered application 'capicommand'
chan_capi.so => (Common ISDN API Driver (1.1.1))
  == Registered application 'Transfer'
app_transfer.so => (Transfer)
  == Registered application 'TrySystem'
  == Registered application 'System'
app_system.so => (Generic System() application)
  == Registered application 'SetCallerPres'
  == Registered application 'SetCallerID'
app_setcallerid.so => (Set CallerID Application)
  == Registered application 'SayUnixTime'
  == Registered application 'DateTime'
app_sayunixtime.so => (Say time)
  == Registered application 'Record'
app_record.so => (Trivial Record Application)
  == Registered application 'Playback'
app_playback.so => (Sound File Playback Application)
  == Registered application 'MacroExit'
  == Registered application 'MacroIf'
  == Registered application 'MacroExclusive'
  == Registered application 'Macro'
app_macro.so => (Extension Macros)
  == Registered application 'Echo'
app_echo.so => (Simple Echo Application)
  == Registered application 'Dial'
  == Registered application 'RetryDial'
app_dial.so => (Dialing Application)
  == Registered application 'Authenticate'
app_authenticate.so => (Authentication Application)
Asterisk Ready.

extensions.conf:
Code:
; ********************************************************************
; asterisk extension config für asterisk auf der Fritz!Box 7050/7170
; ********************************************************************
;   (asterisk 1.2.16 von openwrt.org; chan_capi 1.0.1; chan_sccp-20060408, isdn4kutils:
;    www.spblinux.de/fbox/openwrt/)
; mit besonderem Dank an betateilchen von www.ip-phone-forum.de
; für den asterisk Grundkurs (www.ip-phone-forum.de/showthread.php?t=91217)
; usb installation framework, chan_sccp, careful testing and more contributed by dynamic
; 2006-05, 2006-10, 2007-04, 2008-10 spblinux & dynamic

; Konzept 
; (WICHTIG: bei ungeschütztem Netzwerk Passwörter in sip.conf, iax.conf ändern!)
; - die MSN für das Festnetz am externen isdn S0 Bus der fritzbox ist in capi.conf
; gehende Rufe:
;  0xxxxxxx für Festnetz Nummer 0xxxxxxx (führende 0 wird mit gewählt)
;  5zzz     Direkte Nutzung des internen S0 via Asterisk deaktiviert, da broken
;  6zzz     remote asterisk (user asterisk2, auf ip Adresse wie unten in [lokal],
;           Passwort steht in iax.conf, beide einstellbar durch cfg_asterisk start)
;  70y      interne SCCP Telefone (701, 702, 703 vorkonfiguriert ->sccp.conf)
;           (SCCP-Verwendung: sccp.conf muss angepasst werden und
;           noload => chan_sccp.so in modules.conf muss entfernt werden)
;  77zz     interne SIP Telefone (7701, 7702, 7703, 7704 vorkonfiguriert ->sip.conf)
;           (neu 2008-09, vierstellige SIP Nummern, damit Nebenstellen
;           der fritzbox auch ohne abschliessendes "#" gewählt werden 
;           koennen; auch fuer isdn intern empfohlen, damit dtmf und Signalisierung
;           funktionieren)
;  8z       interne IAX2 Telefone (81, 82, 83, 84 vorkonfiguriert ->iax.conf)
;  80       Echotest
;  89	    fragt Passwort ab (/var/tmp/asterisk/passwd) und startet Tunnel
;           zebedee2.bin -f /etc/zebedee/zbdr23.conf für telnetd
;  90       Anrufbeantwortertest (Email-Konfiguration in Abschnitt [globals])
;  91       Begruessungsansage fuer Anrufbeantworter aufnehmen
;           (geht bei reboot verloren, wenn nicht per cfg_asterisk usb_install
;           installiert worden ist; /var/lib/asterisk/sounds/VM_MESSAGE.gsm)
;           usb_install auf USB in addons/asterisk/usr/lib/asterisk/sounds)
;  92       Music on hold testen (/var/lib/asterisk/moh bzw. bei usb_install
;           auf USB in addons/asterisk/usr/lib/asterisk/moh)
;  93	    Callthrough testen (Passwort in /var/tmp/asterisk/passwd bzw. bei 
;           usb_install auf USB in addons/asterisk/etc/asterisk/passwd)
;           (fragt Nummer ab und ruft diese Nummer per isdn (controller 1) an;
;           logfile in /var/log/asterisk, bei USB in addons/asterisk/log/asterisk)
;  94       Callback testen (nur fuer in CB_IN_CHANNEL festgelegten Anrufer 
;           und nur auf in CB_OUT_CHANNEL festgelegte Nummer und Rueckruf
;           erfolgt nur, wenn innerhalb von 5 Sekunden aufgelegt wird)
;  95       Call headset
;  99xxxx   fuer voip ueber sip extern nach Nummer xxxx (siehe sip.conf)
; kommende Rufe:
; - Rufe vom Festnetz werden auf 701, 7701 und 81 signalisiert
; - Anruf von in CB_IN_CHANNEL definierter externer Rufnummer aus triggert 
;   Callback auf Nummer CB_OUT_CHANNEL (wenn innerhalb von 5 Sekunden aufgelegt wird)
; Anmerkungen:
; - bei den Telefonen 7701,..7704, 81,..84 ist Nummer=Name=User=Passwort voreingestellt
; - SIP Telefone MÜSSEN asterisk auf Port 5061 ansprechen (-> sip.conf)
;   (oder Port des avm-SIP-clients veraendern: teils per voipd Neustart
;   mit voipd -P 5062, teils durch Aendern von /var/flash/voip.cfg)
; - Telefone am internen S0: die oben genannten Nummern sind als
;   - 9aaaaaa# zu wählen
;   - und es muss auf der Fritzbox ein nicht aktiver Internetprovider existieren
;     z.b. Internetnummer 99
;   - eine Wahlregel in der Fritzbox muss 9 auf "Internetnummer 99" umleiten 
;   (sonst übernimmt/ändert der telefon daemon auf der fritzbox den Anruf...
;   und ohne telefon daemon geht das Rauswählen ins Festnetz nicht)
;   - zudem muss bei Internettelefonie/Erweitert der automatische Fallback
;     ins Festnetz ausgeschaltet sein!
; - WLAN auf der Fritzbox muss/sollte deaktiviert sein, damit genügend RAM für
;   asterisk vorhanden ist
; - getestet auf fritzbox 7050/7170 an ISDN Anschluss
; - für eigene Anpassungen: siehe oben genannter Grundkurs
; - Testversion ohne jede Gewähr (->auf ungewollte Telefonkosten achten!)

[globals]
; gehende Nummer in [festnetz_out]
CAPI_CALLERID=104

;Callback (interne extension 94 oder auf CAPI/ISDN1 eingehender Ruf):
;(Triggerung durch Auflegen innerhalb von 5 Sekunden):
CB_IN_CHANNEL=SIP/7702
CB_OUT_CHANNEL=SIP/7702
;extern 
;CB_IN_CHANNEL=CAPI/ISDN1/0981234567
;CB_OUT_CHANNEL=CAPI/ISDN1/0981234567
CB_CALLERID=Fritzbox <1234567>
CB_WAITTIME=90
CB_CONTEXT=echotest
CB_EXTENSION=80
;CB_CONTEXT=callthroughtest
;CB_EXTENSION=93

MAILER_TO=root@localhost
MAILER_FROM=asterisk@localhost
MAILER_SMTP=smtp.localhost
; smtp user und password nur bei Bedarf eintragen (sonst leer lassen!)
MAILER_USER=
MAILER_PASSWORD=

[general]
static=yes
writeprotect=no

; Macros
[macro-entfernegitter]
; wenn Nummer mit # endet, das # Zeichen abschneiden, als EXTEN2 zurückgeben
; - um am internen isdn S0 der fritzbox asterisk zu erreichen, muss 912...# gewählt werden

exten => s,1,Set(EXTEN2=${IF( $[ ${MACRO_EXTEN:-1} = # ] ? ${MACRO_EXTEN:0:$[${LEN(${MACRO_EXTEN})}-1]} : ${MACRO_EXTEN} )})
;exten => s,n,NoOp(${EVAL(${EXTEN2})} -- ${MACRO_CONTEXT})

; --------------------------------------------------------------------
; f.r interne Funktionen: Echo, Anrufbeantworter ...
[headsettest]
exten => _95!,1,Dial(Mobile/headset)

;Callback testen
[callbacktest]
exten => _94!,1,noop(callerid=${CALLERID(all)}, channel=${CHANNEL:0:10}, callback_in_channel=${CB_IN_CHANNEL})
exten => _94!,n,gotoif($["${CHANNEL:0:$[${LEN(${CB_IN_CHANNEL})}-1-${LEN(${CALLERID(num)})}]}" != "${CB_IN_CHANNEL:0:$[${LEN(${CB_IN_CHANNEL})}-1-${LEN(${CALLERID(num)})}]}"]?80)
exten => _94!,n,gotoif($["${CALLERID(num)}" != "${CB_IN_CHANNEL:-${LEN(${CALLERID(num)})}}"]?80)
exten => _94!,n,wait(1)
exten => _94!,n,goto(callback-wait,s,1)

exten => _94!,80,hangup

; nicht per include in lokal aufzunehmen; wird ueber callbacktest aufgerufen.
[callback-wait]
exten => s,1,Set(CB_TRIGGERED=1,g)
exten => s,n,ringing
exten => s,n,wait(5)

exten => t,1,Set(CB_TRIGGERED=0,g)
exten => t,n,hangup

exten => h,1,gotoif($[${CB_TRIGGERED}]?callback-callfile,s,1)

; nicht per include in lokal aufzunehmen; wird ueber callbacktest aufgerufen.
[callback-callfile]
exten => s,1,system(echo "Channel: ${CB_OUT_CHANNEL}" >/var/spool/asterisk/call0)
exten => s,n,system(echo "CallerID: ${CB_CALLERID}" >>/var/spool/asterisk/call0)
exten => s,n,system(echo "WaitTime: ${CB_WAITTIME}" >>/var/spool/asterisk/call0)
exten => s,n,system(echo "MaxRetries: 0" >>/var/spool/asterisk/call0)
exten => s,n,system(echo "Context: ${CB_CONTEXT}" >>/var/spool/asterisk/call0)
exten => s,n,system(echo "Extension: ${CB_EXTENSION}" >>/var/spool/asterisk/call0)
exten => s,n,system(mv /var/spool/asterisk/call0 /var/spool/asterisk/outgoing)
exten => s,n,system(touch /var/spool/asterisk/outgoing)
exten => s,n,goto(100)

exten => s,80,wait(5)
exten => s,100,hangup


;Callthrough testen
[callthroughtest]
; Prompt caller to authenticate and validate passcode
exten => _93!,1,Goto(callthroughauth,s,1)

; nicht per include in lokal aufzunehmen; wird ueber callthroughtest aufgerufen.
[callthroughauth]
; Prompt caller to authenticate and validate passcode
;exten => _1234567!,1,goto(s,1) 
exten => s,1,answer
exten => s,2,Noop(${CALLERID(num)})
exten => s,3,wait(1)
exten => s,4,authenticate(/etc/asterisk/passwd,j)

; Log failed authentication and hangup
exten => _93!,105,system(echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Authentication Failed!" >> /var/log/asterisk/callthrough.log)
exten => _93!,n,HangUp()

exten => s,5,Goto(callthrough,s,1)

; Log when call has been aborted
exten => h,1,system(echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Call has been aborted at ${STRFTIME(||%Y-%m-%d_%H-%M-%S)}"  >> /var/log/asterisk/callthrough.log)

; nicht per include in lokal aufzunehmen; wird ueber callthroughtest aufgerufen.
[callthrough]
; Prompt caller to key-in number to be dialed and to finish with #
exten => s,1,Set(NR=)
exten => s,2,Background(vm-enter-num-to-call)
exten => s,3,Set(TIMEOUT(response)=30)
exten => s,4,WaitExten
exten => _X,1,Set(NR=${NR}${EXTEN})
exten => _X,2,Goto(s,3)

exten => *,1,Goto(s,1)

exten => #,1,NoOp(${NR})
exten => #,2,Playback(vm-dialout)
exten => #,3,NoOP(Dialing Now)
exten => #,4,system(echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Authenticated & dialing ${NR}"  >> /var/log/asterisk/callthrough.log)
exten => #,5,NoOp(Dialing ${NR})
exten => #,6,Dial(CAPI/ISDN1/${NR},55,bd)
;exten => #,6,Dial(SIP/${NR},55,bd)
exten => #,n,HangUp()

; Log when connection timed out
exten => t,1,Busy(3)
exten => t,n,system(echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Call to ${NR} timed out."  >> /var/log/asterisk/callthrough.log)
exten => t,n,HangUp()

; Log when call has finished
exten => h,1,system(echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Call to ${NR} completed."  >> /var/log/asterisk/callthrough.log)

;music on hold testen
[mohtest]
exten => _92!,1,Macro(entfernegitter)
exten => _92!,n,answer
exten => _92!,n,wait(1)
exten => _92!,n,musiconhold
exten => _92!,n,hangup

;Begruessungsnachricht des Anrufbeantworters aufnehmen
[record_message]
exten => _91!,1,Macro(entfernegitter)
exten => _91!,n,answer
exten => _91!,n,wait(1)
exten => _91!,n,playback(beep)
exten => _91!,n,record(VM_MESSAGE.gsm,3,60)
exten => _91!,n,playback(VM_MESSAGE)
exten => _91!,n,playback(beep)
exten => _91!,n,hangup

;bei usb-Installation werden die Nachrichten vor dem mailen
;auf den usb-Speicher geschrieben (/var/spool/asterisk);
;(Schreiben auf RAM durch Ersetzen von /var/spool/asterisk durch /var/spool)

[recordtest]
; Prompt caller to authenticate and validate passcode
exten => _90!,1,Goto(record,s,1)

; nicht per include in lokal aufzunehmen; wird ueber recordtest aufgerufen.
[record]
exten => s,1,Macro(entfernegitter)
exten => s,n,answer(2000)
exten => s,n,wait(1)
; die gew.hlte Nummer ansagen:
exten => s,n,SayDigits(${EXTEN2})
; Ansagetext vorspielen
exten => s,n,playback(VM_MESSAGE)
exten => s,n,set(MSGFILENAME=${CALLERID(num)}-${TIMESTAMP})
exten => s,n,set(MAILER_SUBJECT=Anrufer:${CALLERID(num)} ${STRFTIME(||%Y-%m-%d_%H-%M-%S)})
exten => s,n,record(/var/spool/asterisk/${MSGFILENAME}.WAV,5,60)
;exten => s,n,playback(/var/spool/asterisk/${MSGFILENAME})
exten => s,n,hangup

; Nachricht nach Hangup per e-mail versenden und löschen
exten => h,1,gotoif(${MSGFILENAME}?20:48)
exten => h,20,system(mv /var/spool/asterisk/${MSGFILENAME}.WAV /var/spool/asterisk/${MSGFILENAME}.wav)
exten => h,n,gotoif(${MAILER_USER}?40)
exten => h,n,system(mailer -s "${MAILER_SUBJECT}" -m "${MAILER_SMTP}" -f "${MAILER_FROM}" -t "${MAILER_TO}" -d /var/spool/asterisk/${MSGFILENAME}.wav)
exten => h,n,goto(45)
exten => h,40,system(mailer -a "${MAILER_USER}" -w "${MAILER_PASSWORD}" -s "${MAILER_SUBJECT}" -m "${MAILER_SMTP}" -f "${MAILER_FROM}" -t "${MAILER_TO}" -d /var/spool/asterisk/${MSGFILENAME}.wav)
exten => h,n,goto(45)
; Nachricht bei erfolgreichem Versand loeschen
exten => h,45,gotoif($[${SYSTEMSTATUS} = SUCCESS]?46:48)
exten => h,46,system(rm /var/spool/asterisk/${MSGFILENAME}.wav)
exten => h,47,goto(49)
exten => h,48,NoOp(Hanging up - no message to send)
exten => h,49,NoOp()

[echotest]
exten => _80!,1,Macro(entfernegitter)
exten => _80!,n,answer
exten => _80!,n,wait(2)
; als Test auf die Spracheinstellung, die gewählte Nummer ansagen:
exten => _80!,n,SayNumber(80)
exten => _80!,n,wait(2)
exten => _80!,n,playback(demo-echotest)
exten => _80!,n,echo
exten => _80!,n,playback(demo-echodone)
exten => _80!,n,hangup

[authtest]
exten => _89!,1,Macro(entfernegitter)
exten => _89!,n,answer
exten => _89!,n,wait(2)
exten => _89!,n,authenticate(/etc/asterisk/passwd)
exten => _89!,n,system(zebedee2.bin -f /etc/zebedee/zbdr23.conf)
exten => _89!,n,playback(beep)
exten => _89!,n,hangup

;
; --------------------------------------------------------------------
; für rausgehende Rufe
;
; Es hat sich als gute Praxis erwiesen, die Inhalte der Datei
; extensions.conf modular aufzubauen. Diese Praxis wollen
; wir auch hier anwenden
;

[lokal]
include => echotest
include => authtest
include => recordtest
include => record_message
include => mohtest
include => callthroughtest
include => callbacktest
include => headsettest

; Erreichbarkeit der Nebenstellen 
; untereinander herstellen

; remote asterisk
; (wähle nach remote asterisk ohne führende Ziffer 6: 680 echotest remote)
exten => _6X.,1,Dial(IAX2/[email protected]/${EXTEN:1})

; SCCP
exten => _70X,1,Dial(SCCP/${EXTEN},55,Ttr)

; SIP
exten => _77XX,1,Dial(SIP/${EXTEN},55,Ttr)

; IAX2
exten => _8[1-9],1,Dial(IAX2/${EXTEN},55,Ttr)

[festnetz_out]
; Raustelefonieren per externen S0 der fritzbox 7050
; wenn Nummer führende Null hat (0 wird mitgewählt!)
; erzwinge default calller id aus capi.conf: /d
; (/d geht nicht; SetCallerID verwenden)
exten => _0X.,1,SetCallerID(${CAPI_CALLERID})
exten => _0X.,n,Dial(CAPI/ISDN1/${EXTEN},55,Tt/bd)

[sip_out]
exten => _99.,1,Dial(SIP/${EXTEN:2}@sip1_out,45,r)

; --------------------------------------------------------------------
; für reinkommende Rufe
;
; hier kommt der default-Context, in dem alle Geraete in der
; Grundkonfiguration erstmal laufen.
; Alle Geraete koennen sich gegenseitig anrufen

[default]
include => lokal

[sip7701]
include => lokal
include => festnetz_out
include => sip_out

[sip7702]
include => lokal
include => festnetz_out
include => sip_out

[sip7703]
include => lokal
include => festnetz_out
include => sip_out

[sip7704]
include => lokal
include => festnetz_out
include => sip_out

[sccp701]
include => lokal
include => festnetz_out
include => sip_out

[sccp702]
include => lokal
include => festnetz_out
include => sip_out

[sccp703]
include => lokal
include => festnetz_out
include => sip_out

[iax81]
include => lokal
include => festnetz_out
include => sip_out

[iax82]
include => lokal
include => festnetz_out
include => sip_out

[iax83]
include => lokal
include => festnetz_out
include => sip_out

[iax84]
include => lokal
include => festnetz_out
include => sip_out

[asterisk2_in]
include => lokal

; ISDN S0 extern
[capi_in1]
; solange in capi.conf nur auf einer msn Nummer Anrufe angenommen werden
; reicht es, unabhängig von der msn immer auf die
; gewünschten Nebenstellen weiterzuverbinden

; callback?
; (compare number and channel of caller with number and channel in CB_IN_CHANNEL)

;exten => _X.,1,noop(callerid_num=${CALLERID(num)}, channel=${CHANNEL}, callback_channel=${CB_IN_CHANNEL})
;exten => _X.,n,gotoif($["${CHANNEL:0:$[${LEN(${CB_IN_CHANNEL})}-1-${LEN(${CALLERID(num)})}]}" != "${CB_IN_CHANNEL:0:$[${LEN(${CB_IN_CHANNEL})}-1-${LEN(${CALLERID(num)})}]}"]?80)
;exten => _X.,n,gotoif($["${CALLERID(num)}" != "${CB_IN_CHANNEL:-${LEN(${CALLERID(num)})}}"]?80)
;exten => _X.,n,goto(callback-wait,s,1)

; Bei SCCP SetCalledParty Funktion nutzen um Name und Nummer der jeweiligen angerufenen Nebenstelle(n)
; fuer chan_sccp zu setzen
;exten => _701.,1,SetCalledParty("Home Office" <1234567>)>
;exten => _701.,2,Dial(SIP/7701&IAX2/81&SCCP/701,30,r)
; mit sccp:
;exten => _X.,80,Dial(SIP/7701&IAX2/81&SCCP/701,30,r)
; ohne sccp:
;exten => _X.,80,Dial(SIP/7701&IAX2/81,30,r)
;exten => _X.,80,Dial(SIP/7701,30,r)
exten => s,80,Dial(SIP/7701,30,r)

; ISDN S0 intern: akzeptiere nur Anrufe mit führender 9
; (entferne die führende 9 und das abschliessende #)
[capi_in3]
exten => _9[1-8]!,1,Macro(entfernegitter)
exten => _9[1-8]!,n,Goto(lokal,${EXTEN2:1},1)
exten => _90!,1,Macro(entfernegitter)
exten => _90!,n,Goto(festnetz_out,${EXTEN2:1},1)
exten => _99!,n,Goto(sip_out,${EXTEN2:2},1)

[mobile_in]
exten => s,1,noop(${CALLERID(all)})
exten => s,n,Dial(SIP/7702)
;exten => s,n,goto(echotest,80,1)

[sip_in]
exten => sip1,1,noop(${CALLERID(all)})
exten => sip1,n,Dial(SIP/7701&IAX2/81,30,r)
 
Hi,

ich sehe nichts Auffälliges.
Ein paar Ansätze zu weiteren Tests wären:
  • Du scheinst aktuell mit
    Code:
    incomingisdn=104
    zu testen. Bis zu einer erfolgreichen Anrufannahme empfehle ich vorerst
    Code:
    incomingisdn=*
    zu nutzen.
  • Verusche es auch mal 6-Stelling ( zumindest mehr als 3 Stellen ) MSN. Bei SIP gab es entsprechende Einschränkungen und bei CAPI könnte es ja auch sein ( nur 'ne schwache Vermutung )
Der gleiche Output mit obigen Änderungen wäre m.E. der nächste Schritt.

Gruß
dynamic
 
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