Problem: Raus nein, Rein ja

[_benjamin]

Hallo,

hinter dem Router T-Sinus 154 DSL habe ich * installiert. Ich kann reintelefonieren. Jedoch nicht raus.

extensons.conf

;Extension Configuration
[general]
static = yes
writeprotect = no

[internsip]
exten => _XX,1,Dial(SIP/${EXTEN},15)
exten => _XX,n,Voicemail(u${EXTEN})

exten => 11,1,Milliwatt()

exten => 50,1,VoicemailMain()

exten => 41,1,DateTime()
exten => 41,n,Hangup

exten => 42,1,Answer()
exten => 42,n,Echo()

exten => _0.,1,Set(CALLERID(name)=032XXXXXXXXX)
exten => _0.,n,Dial(SIP/${EXTEN}@tel.t-online.de,60,tr)
exten => _0.,n,Congestion

[ankommend]
exten => 032XXXXXXXX,1,Dial(SIP/30)

sip.conf

;SIP Configuraton

[general]
context = voip_in
port = 5060
bindaddr = 0.0.0.0
language = de
externsip = XXX.dyndns.org
srvlookup = yes
localnet = 192.168.0.0/24
maxexpirey = 240
defaultexpirey = 240

register => 032XXXXXXXX:password:[email protected]/032XXXXXXX 
#include sip_benutzer.conf

sip_benutzer.conf

;Benutzerdatei SIP Nutzer

[30]
username = 30
secret = 30
type = friend
qualify = yes
nat = no
host = dynamic
context = internsip
canreinvite = no

[t-online_out]
username = emailadresse
authuser = emailadresse
secret = password
type = user
host = tel.t-online.de 
fromuser = 032XXXXXXX
fromdomain = tel.t-online.de
canreinvite = no 
qualify = yes 
disallow = all
allow = ulaw
realm = tel.t-online.de
insecure = very
nat = yes
dtmfmode = info
context = internsip 
canreinvite = no

[t-online_in]
type = peer
fromdomain = tel.t-online.de
host = tel.t-online.de
disallow = all
allow = ulaw
nat = yes
context = ankommend

Vielen Dank für jede Hilfe!

Mit freundlichen Grüßen


Benjamin

 

[betateilchen]

exten => _0.,n,Dial(SIP/${EXTEN}@[COLOR="Red"]tel.t-online.de[/COLOR],60,tr)

Ersetze mal das oben rot markierte durch t-online_out :wink:

 

[_benjamin]

Genauso hatte ich es einmal. Habe es aber wieder verworfen, weil es folgenden Fehler brachte:

Apr 15 14:35:53 DEBUG[2646] chan_sip.c: Checking SIP call limits for device 30
Apr 15 14:35:53 DEBUG[2646] chan_sip.c: build_route: Contact hop: <sip:[email protected]>
Apr 15 14:35:54 WARNING[2660] chan_sip.c: No such host: t-online_out
Apr 15 14:35:54 NOTICE[2660] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Apr 15 14:35:54 DEBUG[2660] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.
Apr 15 14:35:54 VERBOSE[2660] logger.c: == Spawn extension (internsip, XXXXXX, 3) exited non-zero on 'SIP/30-a50d'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '"032221207878" <30>'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '30'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is 'XXXXXX'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is 'internsip'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is 'SIP/30-a50d'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '(null)'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is 'Congestion'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '(null)'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '2006-04-15 14:35:53'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '(null)'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '2006-04-15 14:35:54'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '1'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '0'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is 'FAILED'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is 'DOCUMENTATION'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '(null)'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '1145104553.2'
Apr 15 14:35:54 DEBUG[2660] pbx.c: Function result is '(null)'

Es scheint Asterisk kann gar keine Verbindung aufbauen. Aber wieso?