[asterisk-users] Error redirecting an incoming call of a SIP provider to a local extension
Daniel Bareiro
daniel-listas at gmx.net
Fri Feb 19 18:35:07 CST 2010
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Hi all!
I am trying to redirect to a local extension the incoming calls that I
receive to an account which I have in iptel.org, but when receiving I'm
obtaining this error:
alderamin*CLI>
-- Executing [300 at from-internal:1] Dial("SIP/danib-089f8820",
"SIP/300|30|tTrm") in new stack
[Feb 19 19:22:50] WARNING[19254]: app_dial.c:1237 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
[Feb 19 19:23:00] WARNING[19254]: pbx.c:2529 __ast_pbx_run: Timeout, but
no rule 't' in context 'from-internal'
It is probable that this can be due to a problem of interaction between
contexts? I copy the content of extensions.conf and sip.conf to see if
it can help to find the problem:
- ------------------------------------------------------------------------
extensions.conf:
; DGB - 20091114
[general]
autofallthrough=no
[macro-dial]
exten => s,1,Dial(${ARG1},15)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@voicemail,u)
exten => s-NOANSWER,n,Hangup
exten => s-BUSY,1,Voicemail(${MACRO_EXTEN}@voicemail,b)
exten => s-BUSY,n,Hangup
exten => s-CHANUNAVAIL,1,Playback(pbx-invalid)
[from-internal]
; Llamadas a extensiones SIP
exten => _2xx,1,Macro(dial,SIP/${EXTEN})
exten => _2xx,n,Hangup
exten => 300,1,Dial(SIP/300,30,tTrm)
; Extension analogica
exten => 402,1,Macro(dial,DAHDI/2)
exten => 402,n,Hangup
; Directorio de extensiones
exten => *400,1,Directory(voicemail,from-internal)
; Musica en espera
exten => *300,1,Answer
exten => *300,n,SetMusicOnHold(default)
exten => *300,n,WaitMusicOnHold(2000)
exten => *300,n,Hangup
; Prueba de Eco
exten => *200,1,Answer
exten => *200,n,Playback(demo-echotest)
exten => *200,n,Echo
exten => *200,n,Playback(demo-echodone)
exten => *200,n,Hangup
; Acceso a voicemail
exten => *100,1,Answer
exten => *100,n,Wait(1)
exten => *100,n,VoiceMailMain(${CALLERID(num)}@voicemail)
exten => *100,n,Hangup
; Llamadas salientes
exten => _9.,1,Dial(DAHDI/1/${EXTEN:1})
exten => _9.,n,Hangup
; Call a number at iptel.org
exten => _0.,1,Dial(SIP/iptel/${EXTEN:1},20,r))
exten => _0.,n,Hangup
[from-pstn]
; incoming calls from FXO port are directed to this context
exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=15)
exten => s,n,Background(contestador1)
exten => i,1,Goto(from-pstn,s,1)
exten => t,1,Playback(locomunicoconelinterno1)
exten => t,n,Dial(SIP/200,25)
exten => t,n,VoiceMail(200 at voicemail,20)
exten => t,n,Hangup()
include => from-internal
- ------------------------------------------------------------------------
sip.conf:
[general]
[...]
; register with iptel.org
register => danib:mLrZvbnb at iptel.org/300
[...]
; Outgoing to iptel.org
[iptel]
type=friend
username=danib
secret=myspasswd
host=iptel.org
canreinvite=no
qualify=300
insecure=port,invite ; required for incoming ekiga.net calls
context = from-internal
- ------------------------------------------------------------------------
Thanks in advance for your replies.
Regards,
Daniel
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