asterisk users - Feb 2010 - Error redirecting an incoming call of a SIP provider to a local extension

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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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On 20/02/2010 01:35, Daniel Bareiro wrote:
> 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)
>    
Well, looks like your * server is simply unable to dial the sip user
'300'.
  Either there is some call-limit in place, or problem with the registration
of the phone ?
> 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:
>    
What could be of some use, is the result of "sip show peer 300"