CAPI zu SIP-Client geht, andersherum nicht

[dERDA05]

Hallo,
nutze Asterisk auf Eisfair.
Wenn ich eine bestimmte externe MSN wähle, soll * zum SIP-Client mit der internen MSN 10 weiterleiten. Funktioniert Prima!
Wenn ich aber vom SIP-Client eine Nummer (ob eine interne TEST-MSN oder eine externe Festnetznummer) wähle, kommt irgendwann ein Timeout.
Ein Calltrough-Script über den VOIP-Provider(bellshare) funktioniert aber!
Server und Client sind im gleichen Subnetz(Server 192.168.1.5, Client 192.168.1.2)
Client hat keine Firewall!

Hier ein Auszug aus "var/log/asterisk/full".

Feb 27 20:48:42 VERBOSE[4090] logger.c:     -- Registered SIP '10' at 192.168.1.2 port 38351 expires 5
Feb 27 20:48:42 DEBUG[4090] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
Feb 27 20:48:43 DEBUG[4090] chan_sip.c: Auto destroying call 'Y2M4OGJiMjZjYzkzMDMyYTAxMzVkN2VjNDQyZDRkYTI.'
Feb 27 20:48:43 VERBOSE[4271] logger.c:     -- Remote UNIX connection disconnected
Feb 27 20:48:52 DEBUG[4090] chan_sip.c: Auto destroying call 'YjliYTdlN2Y2NzUwYWRjOTJhNTY4OWM3NjRiYWUwODE.'

/var/log/asterisk/messages

Feb 27 20:46:22 NOTICE[4090] chan_sip.c: Peer '10' is now UNREACHABLE!  Last qualify: 0
Feb 27 20:46:22 NOTICE[4090] chan_sip.c: Peer '10' is now REACHABLE! (2ms / 1000ms)
Feb 27 20:52:18 NOTICE[8512] cdr.c: CDR simple logging enabled.
Feb 27 20:52:18 WARNING[8512] res_musiconhold.c: Unable to open pseudo channel for timing...  Sound may be choppy.
Feb 27 20:52:18 WARNING[8512] chan_iax2.c: Unable to open IAX timing interface: No such device

sip.conf:

[general]
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
context=default
defaultexpirey=120
maxexpirey=3600
disallow=all
allow=ulaw
localnet=192.168.1.0/24
canreinvite=yes
tos=reliability
nat=yes
dtmfmode=auto
qualify=no
language=de

; Write Registries
register => "ich":"mein_passwort"@bellshare.com/"meine externe MSN"

; Write SIP-Proxy accounts
[bellshare]
type=friend
host=bellshare.com
fromuser="ich"
username="ich"
secret="mein_passwort"
fromdomain=bellshare.com
context=incoming
insecure=very
disallow=all
allow=ulaw
nat=no

; Write SIP-Client accounts
[10]
type=friend
secret=1234
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=auto
disallow=all
allow=ulaw
context=outgoing

Dann gibts noch die extensions.conf:

[general]

static=yes
writeprotect=no

[globals]
localdialtimeout=120

[default]
include => user
include => test
include => incoming
include => outgoing
include => error

[test]
exten => #100,1,Set(LANGUAGE()=de)
exten => #100,2,Answer()
exten => #100,3,MP3Player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3)
exten => #100,4,Congestion()
exten => #100,5,Wait(10)
exten => #100,6,Hangup()

exten => #101,1,Set(LANGUAGE()=de)
exten => #101,2,Answer()
exten => #101,3,Milliwatt()
exten => #101,4,Congestion()
exten => #101,5,Wait(10)
exten => #101,6,Hangup()

exten => #102,1,Set(LANGUAGE()=de)
exten => #102,2,Answer()
exten => #102,3,SayDigits(${CALLERIDNUM})
exten => #102,4,Congestion()
exten => #102,5,Wait(10)
exten => #102,6,Hangup()

exten => #103,1,Set(LANGUAGE()=de)
exten => #103,2,Answer()
exten => #103,3,Playback(demo-echotest)
exten => #103,4,Echo()
exten => #103,5,Playback(demo-echodone)
exten => #103,6,Congestion()
exten => #103,7,Wait(10)
exten => #103,8,Hangup()

exten => #104,1,Set(LANGUAGE()=de)
exten => #104,2,Answer()
exten => #104,3,AGI(date.sh)
exten => #104,4,SendText(${ADATE})
exten => #104,5,sayPhonetic(${ADATE:5})
exten => #104,6,Congestion()
exten => #104,7,Wait(10)
exten => #104,8,Hangup()

[outgoing]
include => 10

include => incoming

[10]
;;; Phone 10 ;;;
; Outgoing calls over SIP
exten => _x.,1,Set(LANGUAGE()=de)
exten => _x./10,2,Ringing()
exten => _x./10,3,Set(CALLERID(number)="extern_msn)
exten => _x./974143,4,Set(CALLERID(name)=)
exten => _x./974143,5,Dial(SIP/${areaCode}${EXTEN}@bellshare,120,TW)
exten => _x./974143,6,Goto(dialStates,${DIALSTATUS},1)


; Outgoing calls over CAPI
exten => _000.,1,Set(LANGUAGE()=de)
exten => _000./10,2,Set(CALLERID(number)="extern_msn)
exten => _000./974143,3,Set(CALLERID(name)=)
exten => _000./974143,4,Dial(CAPI/contr1/${EXTEN:3}/b,120,TW)
exten => _000./974143,5,Goto(dialStates,${DIALSTATUS},1)



[incoming]
include => default

exten => 974143,1,Set(LANGUAGE()=de)
exten => 974143,2,Ringing()
exten => 974143,3,Dial(SIP/10,${localdialtimeout},t)
exten => 974143,4,Congestion()
exten => 974143,5,Wait(10)
exten => 974143,6,Hangup()

; Incoming (internal) calls for number 10. Type of phone: sip
exten => *10,1,Set(LANGUAGE()=de)
exten => *10,2,NoOp()
exten => *10,3,NoOp()
exten => *10,4,Dial(SIP/10,120)
exten => *10,5,Goto(incoming,*10-${DIALSTATUS},1)
exten => *10,6,Goto(dialStates,${DIALSTATUS},1)


[dialStates]
exten => NOANSWER,1,Set(LANGUAGE()=de)
exten => NOANSWER,2,Playback(vm-nobodyavail)
exten => NOANSWER,3,Congestion()
exten => NOANSWER,4,Wait(10)
exten => NOANSWER,5,Hangup()

exten => BUSY,1,Set(LANGUAGE()=de)
exten => BUSY,2,Busy()
exten => BUSY,3,Congestion()
exten => BUSY,4,Wait(10)
exten => BUSY,5,Hangup()

exten => CHANUNAVAIL,1,Set(LANGUAGE()=de)
exten => CHANUNAVAIL,2,Congestion()
exten => CHANUNAVAIL,3,Congestion()
exten => CHANUNAVAIL,4,Wait(10)
exten => CHANUNAVAIL,5,Hangup()

exten => CONGESTION,1,Set(LANGUAGE()=de)
exten => CONGESTION,2,Congestion()
exten => CONGESTION,3,Congestion()
exten => CONGESTION,4,Wait(10)
exten => CONGESTION,5,Hangup()


[error]
exten => h,1,Hangup
exten => _[*#X].,1,Set(LANGUAGE()=de)
exten => _[*#X].,2,Playback(error)
exten => _[*#X].,3,Playback(pbx-invalid)
exten => _[*#X].,4,Congestion()
exten => _[*#X].,5,Wait(10)
exten => _[*#X].,6,Hangup()

[user]

Ich hoffe, ihr könnt mir helfen!

Dennis

 

[dERDA05]

Ich hab den Fehler immer noch nicht gefunden.
Hab aus Frust eisfair hinter mir gelassen .
Hab jetzt Debian drauf und kompiliere mir mein Asterisk. Mal sehen was rauskommt ...