ASterisk auf Vserver mit G 729 Support

[rmh]

Für mich sieht das so aus, als ob mindestens ein include fehlt.
Setze mal in deine extensions.conf sinngemäß

[von-voip-provider]
include => meine-telefone

...

Gruß
R.

 

[koepie]

Hat leider auch nichts gebracht.

* Hier noch mal aktuelle konfigs
sip

[general]
context = default
srvlookup=yes
port = 5060
bindaddr = 78.47.249.xx
language=de


register => 4936453893xx:[email protected]/4936453893xx




[2000]
type=friend
context = meine-telefone
secret=xxxx
;qualify=no    ; Qualify peer is not more than 2000 mS away
nat=yes         ; This phone is natted
host=dynamic   ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect



[2001]
type=friend
context = meine-telefone
secret=xxxxx
qualify=no    ; Qualify peer is not more than 2000 mS away
nat=yes         ; This phone is natted
host=dynamic   ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect



[ext-sip-account]
type=friend
context = income
username=4936453893xx
fromuser=4936453893xx
secret=xxxxx
host=sip.1und1.de
fromdomain=sip.1und1.de
nat=yes

extension

[default]
[meine-telefone]
; ext 2: echotest
exten => 2,1,Answer()
exten => 2,n,Wait(2)
exten => 2,n,Playback(demo-echotest) ; Let them know what
exten => 2,n,Echo()                  ; Do the echo test
exten => 2,n,Playback(demo-echodone) ; Let them know it
exten => 2,n,Hangup()


; ext 4:  the speaking clock
exten => 4,1,Answer
exten => 4,n,Wait(2)
exten => 4,n,SayUnixTime( | | k)
exten => 4,n,SayUnixTime( | | M)
;exten => 4,n,Playback(vm-and)
;exten => 4,n,SayUnixTime( | | S)
;exten => 4,n,Playback(seconds)
exten => 4,n,Hangup




exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,VoiceMail(2000,u)

exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,VoiceMail(2001,u)

exten => 2999,1,VoiceMailMain(${CALLERID(num)},s)

exten => _0[1-9].,1,Dial(SIP/${EXTEN}@ext-sip-account)

[income]
include => meine-telefone
exten => 4936453893xx,1,Dial(SIP/2000)

Immer noch die selbe Fehlermeldung

[Jun 16 11:17:47] NOTICE[21935]: chan_sip.c:20039 handle_request_invite: Call from ' ' to extension '4936453893' rejected because extension not found.

An der Fehlermeldung finde ich halt auch komisch warum steht da from ' ' und nichts drin ?

 

[rmh]

Machst du ein "dialplan reload" wenn du an der extensions.conf etwas änderst?

PS: Versuche ein core set verbose 10 und schau dir auf dem CLI an, welcher Kontext während des Anrufversuchs verwendet werden.

-- Executing [[I]nebenstelle[/I]@[B][I]kontext[/I][/B]:[I]prioriät[/I]]

Gruß
R.

 

[koepie]

So hab mal verbose 10 gesetzt da kommt leider auch nicht mehr raus :/

 dialplan reload
Dialplan reloaded.
  == Parsing '/etc/asterisk/extensions.conf':   == Found
    -- Registered extension context 'default' (0x219a9af8) in local table 0x21ca                                      8fc8; registrar: pbx_config
    -- Registered extension context 'meine-telefone' (0x219abdd8) in local table                                       0x21ca8fc8; registrar: pbx_config
    -- Added extension '2' priority 1 to meine-telefone (0x219abdd8)
    -- Added extension '2' priority 2 to meine-telefone (0x219abdd8)
    -- Added extension '2' priority 3 to meine-telefone (0x219abdd8)
    -- Added extension '2' priority 4 to meine-telefone (0x219abdd8)
    -- Added extension '2' priority 5 to meine-telefone (0x219abdd8)
    -- Added extension '2' priority 6 to meine-telefone (0x219abdd8)
    -- Added extension '4' priority 1 to meine-telefone (0x219abdd8)
    -- Added extension '4' priority 2 to meine-telefone (0x219abdd8)
    -- Added extension '4' priority 3 to meine-telefone (0x219abdd8)
    -- Added extension '4' priority 4 to meine-telefone (0x219abdd8)
    -- Added extension '4' priority 5 to meine-telefone (0x219abdd8)
    -- Added extension '2000' priority 1 to meine-telefone (0x219abdd8)
    -- Added extension '2000' priority 2 to meine-telefone (0x219abdd8)
    -- Added extension '2001' priority 1 to meine-telefone (0x219abdd8)
    -- Added extension '2001' priority 2 to meine-telefone (0x219abdd8)
    -- Added extension '2999' priority 1 to meine-telefone (0x219abdd8)
    -- Added extension '_0[1-9].' priority 1 to meine-telefone (0x219abdd8)
    -- Registered extension context 'income' (0x219aa230) in local table 0x21ca8                                      fc8; registrar: pbx_config
    -- Including context 'meine-telefone' in context 'income'
    -- Added extension '4936453893xx' priority 1 to income (0x219aa230)
  == Parsing '/etc/asterisk/users.conf':   == Found
    -- Registered extension context 'app_dial_gosub_virtual_context' (0x219ac0c0                                      ) in local table 0x21ca8fc8; registrar: app_dial
    -- merging incls/swits/igpats from old(app_dial_gosub_virtual_context) to ne                                      w(app_dial_gosub_virtual_context) context, registrar = pbx_config
    -- Added extension 's' priority 1 to app_dial_gosub_virtual_context (0x219ac                                      0c0)
    -- Registered extension context 'parkedcalls' (0x21c67260) in local table 0x                                      21ca8fc8; registrar: features
    -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) cont                                      ext, registrar = pbx_config
    -- Added extension '700' priority 1 to parkedcalls (0x21c67260)
    -- Registered extension context 'app_queue_gosub_virtual_context' (0x21cab38                                      0) in local table 0x21ca8fc8; registrar: app_queue
    -- merging incls/swits/igpats from old(app_queue_gosub_virtual_context) to n                                      ew(app_queue_gosub_virtual_context) context, registrar = pbx_config
    -- Added extension 's' priority 1 to app_queue_gosub_virtual_context (0x21ca                                      b380)
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000034                                       sec
    -- Time to restore hints and swap in new dialplan: 0.000001 sec
    -- Time to delete the old dialplan: 0.000007 sec
    -- Total time merge_contexts_delete: 0.000042 sec

---> Das kommt auf der CLI wenn ich anrufe

  == Using SIP RTP CoS mark 5
[Jun 16 12:27:45] NOTICE[21935]: chan_sip.c:20039 handle_request_invite: Call from '' to extension '4936453893xx' rejected because extension not found.

*Ich habe mitlerweile sogar das Asterisk buch durchgelsen und auch verstanden.
Aber ich verstehe nicht warum diese einfach Konfiguration nicht funktioniert?!