Another help in english-Asterisk, 7170 and analog extesioons

Hi to everybody,
most probably the question I have is already here but is not easy to understand german, sorry.
I have done many trials recently and everything was working ok, except for one point.
I have now installed the lastest asterisk on 7170 with 04.47 on usb.
Ther configs are the basic ones.
Sip phones, sip registration, call, incoming sip calls, etc are all ok.
Where I cannot go on is the analog pst lines.
I cannot get them working either out or in.
On my capi.conf I tried on ISDN1 to use controller 1 or 4 with no difference.
There is no sign of activity on the cli console when a call come in the analog line and I get:

-- Executing [0574633080@sip771:2] Dial("SIP/771-005f0d28", "CAPI/ISDN1/057*******|55|Tt/bd") in new stack
-- Called ISDN1/057*******
-- CAPI/ISDN1#02/057*******-0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/771-005f0d28' status is 'CONGESTION'

When I try to use outgoing call on analog line.

Can anybody be so kind to tell me what I have missed ??
- I have set up the 3 internet number to be used by the 3 analog line of S0 and registered on asterisk.
- On internet telephony --> advanced, Use fixed-line substitute connection is not clikked.
- The 3 S0 extensions are registered on the 3 internet numbers above.
- I tried also also to register the fixed line to the number 772 with no result.

Thanks in advance.
Enrico

This is the capi.conf part

[ISDN1]
ntmode=no
isdnmode=msn
incomingmsn=057*******
defaultcid=057*******
controller=4
group=1
softdtmf=off
relaxdtmf=off
accountcode=
context=capi_in1
bridge=no
devices=2
immediate=yes

this is the extensions.conf part:
[festnetz_out]
exten => _0X.,1,SetCallerID,${CAPI_CALLERID}
exten => _0X.,n,Dial,CAPI/ISDN1/${EXTEN}|55|Tt/bd

[capi_in1]
exten => _X.,80,Dial,SIP/771&IAX2/81&SCCP/701&CAPI/ISDN3/31|30|r

Any additional set up in FB to let this work ??
Many thanks for your reply.
Enrico

This is exactly what I was doing, but nothing !
Still there is no activity on the CLI console !
What about this:

- Telefone am internen S0: die oben genannten Nummern sind als
- 9aaaaaa# zu wõhlen
- und es muss auf der Fritzbox ein nicht aktiver Internetprovider existieren
z.b. Internetnummer 99
- eine Wahlregel in der Fritzbox muss 9 auf "Internetnummer 99" umleiten
(sonst ³bernimmt/õndert der telefon daemon auf der fritzbox den Anruf...
und ohne telefon daemon geht das Rauswõhlen ins Festnetz nicht)
- zudem muss bei Internettelefonie/Erweitert der automatische Fallback
ins Festnetz ausgeschaltet sein!

I really do not understand it !
Is something that concern my problem ?? (if not do not spend time to explain what it is !!!)
Thanks
Enrico

Changing the extensions with your example, dialing out now I get this:

-- Executing [057*****@sip771:1] Dial("SIP/771-005e1390", "CAPI/ISDN 1/23:0
574633080|55|T/bdr") in new stack
-- didn't find capi device for interface 'ISDN1'

[Jan 20 20:32:24] WARNING[1879]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'CAPI' (cause 44 - Requested channel not available)
Unable to create channel of type 'CAPI' (cause 44 - Requested channel not available)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/771-005e1390' status is 'CHANUNAVAIL'

Seems to me that * is not intercepting at all the capi of the FB.
If I assign in FB Telephony-->extensions-->Fon 1 registered as 771 on *, the fix line as additional numer, clearly the phone ring, as the call is handled from the FB and not *, and this means to me that the hardware is ok.
I still have a big suspect that I miss something in the config that I have not understood.

Enrico

Hi Tippfehler
yes I did change devices=1
but I did not put bridge=yes
And THIS was the problem !!!

Now evething is working !!!
Thank you really very much for your help !!!

Enrico