keine ausgehenden Anrufe nöglich

OhOme

Neuer User
Mitglied seit
15 Jan 2009
Beiträge
7
Punkte für Reaktionen
0
Punkte
0
hallo,

ich versuche nun schon seit einigen Stunden, mit meinem *
über 1und1 rauszuwählen. hoffentlich kann mir jemand helfen - ich seh den wald vor lauter bäumen nicht mehr. :confused:
intern funktioniert alles, beim rauswählen kommt immer die fehlermeldung:
[Jan 29 18:53:27] WARNING[2034]: pbx.c:3082 pbx_extension_helper: No application 'Dial,SIP/${EXTEN}@492831234567|45|r' for extension (sip7702, 04411, 1)

== Spawn extension (sip7702, 04411, 1) exited non-zero on 'SIP/7702-0062e7b8'

Meine sip.conf

[Edit frank_m24: Bitte benutzt CODE Tags für solche Ausgaben.]
Code:
---------------------------------------------------------------------
[general]
context=default                 ; Default context for incoming calls
bindport=5061                   ; UDP Port to bind to (SIP standard port is 5060
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
language=de
maxexpirey=3600
defaultexpirey=1800
externhost=porrmann.dyndns.org



localnet=192.168.0.0/255.255.255.0
nat=yes
register => 492831234567:[email protected]/492831234567

[7701]
context=sip7701
callerid="TestSIP 7701" <7701>
host=dynamic
domain=192.168.0.254
;nat=yes
qualify=no                     ; X-Lite is behind a NAT router
type=friend
user=7701
secret=7701
disallow=all
allow=gsm                     ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw

[7702]
context=sip7702
callerid="TestSIP 7702" <7702>
host=dynamic
domain=192.168.0.254
type=friend
user=7702
secret=7702
disallow=all
allow=gsm                     ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw

...

; sip external outgoing
[492831234567]
type=peer
username=492831234567
fromuser=492831234567
secret=geheim
host=sip.1und1.de
fromdomain=1und1.de
insecure=invite
canreinvite=no
nat=no
disallow=all
allow=ulaw

Meine extensions.conf

---------------------------------------------------------------------

Code:
[globals]

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

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

[sip_out]
exten => _0.,1,Dial,SIP/${EXTEN}@492831234567|45|r

[default]
include => lokal
include => sip_out
include => record-outboundmsgs

[sip7701]
include => lokal
include => sip_out
include => record-outboundmsgs

[sip7702]
include => lokal
include => sip_out
include => record-outboundmsgs

[sip7703]
include => lokal
include => sip_out
include => record-outboundmsgs

[sip7704]
include => lokal
include => record-outboundmsgs
include => sip_out

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

[outboundmsg1]
exten => s,1,Set(TIMEOUT(digit)=5)             ; Set Digit Timeout to 5 seconds
exten => s,2,Background(outboundmsgs/msg1)         ; "play outbound msg"
exten => s,3,ResponseTimeout(10)        ; Set Response Timeout to 10 seconds
exten => s,4,Answer
exten => s,5,Wait(1)
exten => s,6,Background(outboundmsgs/msg1)         ; "play outbound msg"
exten => s,7,Background(outboundmsgs/how_to_ack)   ; "Press 1 to replay or 2 to
exten => 1,1,Goto(s,5)   ; replay message
exten => 2,1,Goto(msgack,s,1) ; acknowledge message
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup
; at this point we could do something like reschedule the call to try again late
; or send an email saying the msg was not received,
; or ...

[outboundmsg2]
exten => s,1,DigitTimeout,5             ; Set Digit Timeout to 5 seconds
exten => s,2,ResponseTimeout,10         ; Set Response Timeout to 10 seconds
exten => s,3,Answer
exten => s,4,Wait(1)
exten => s,5,Background(outboundmsgs/msg2)         ; "play outbound msg"
exten => s,6,Background(outboundmsgs/how_to_ack)   ; "Press 1 to replay or 2 to
exten => 1,1,Goto(s,5)   ; replay message
exten => 2,1,Goto(msgack,s,1) ; acknowledge message
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup
; at this point we could do something like reschedule the call to try again late
; or send an email saying the msg was not received,
; or ...

[msgack]
exten => s,1,Playback(outboundmsgs/thankyou)
exten => s,2,Playback(vm-goodbye)
exten => s,3,Hangup
; at this point we might want to log the message acknowledgement somewhere
; and perhaps trigger some additional processing

[record-outboundmsgs]
; outbound msg1
exten => 2051,1,Wait(2)
exten => 2051,2,Record(outboundmsgs/msg1:gsm)
exten => 2051,3,Wait(2)
exten => 2051,4,Playback(outboundmsgs/msg1)
exten => 2051,5,wait(2)
exten => 2051,6,Hangup
;
; outbound msg2
exten => 2052,1,Wait(2)
exten => 2052,2,Record(outboundmsgs/msg2:gsm)
exten => 2052,3,Wait(2)
exten => 2052,4,Playback(outboundmsgs/msg2)
exten => 2052,5,wait(2)
exten => 2052,6,Hangup

; Msg played when msg is acked
exten => 2061,1,Wait(2)
exten => 2061,2,Record(outboundmsgs/thankyou:gsm)
exten => 2061,3,Wait(2)
exten => 2061,4,Playback(outboundmsgs/thankyou)
exten => 2061,5,wait(2)
exten => 2061,6,Hangup
;
; Msg played after outbound msg: "Press 1 to replay or 2 to acknowledge receivin
exten => 2062,1,Wait(2)
exten => 2062,2,Record(outboundmsgs/how_to_ack:gsm)
exten => 2062,3,Wait(2)
exten => 2062,4,Playback(outboundmsgs/how_to_ack)
exten => 2062,5,wait(2)
exten => 2062,6,Hangup
 
Du scheinst unter
Code:
[sip_out]
exten => _0.,1,Dial,SIP/${EXTEN}@492831234567|45|r
die "alte" Schreibweise zu nutzen, die nicht mehr gültig ist.
Versuche es mal mit:
Code:
[sip_out]
 exten => _0.,1,Dial(SIP/${EXTEN}@492831234567,45,r)
Gruß
dynamic
 
welche asterisk version hast Du?
In 1.6 hat sich was geändert, da funtionieren die | Zeichen nicht mehr.

Im Internen hast Du die Klammern gesetzt, mach das auch mal bei Extern:
[sip_out]
exten => _0.,1,Dial(SIP/${EXTEN}@492831234567,45,r)
 
Oh Sorry

dynamic war schneller
 
THX

das wars. DANKE
 

Zurzeit aktive Besucher

Neueste Beiträge

Statistik des Forums

Themen
244,878
Beiträge
2,220,027
Mitglieder
371,604
Neuestes Mitglied
broekar
Holen Sie sich 3CX - völlig kostenlos!
Verbinden Sie Ihr Team und Ihre Kunden Telefonie Livechat Videokonferenzen

Gehostet oder selbst-verwaltet. Für bis zu 10 Nutzer dauerhaft kostenlos. Keine Kreditkartendetails erforderlich. Ohne Risiko testen.

3CX
Für diese E-Mail-Adresse besteht bereits ein 3CX-Konto. Sie werden zum Kundenportal weitergeleitet, wo Sie sich anmelden oder Ihr Passwort zurücksetzen können, falls Sie dieses vergessen haben.

IPPF im Überblick

Neueste Beiträge