asterisk users - Mar 2004 - Guru's help with * and AVM C2 ISDN - Newbie going mad!!

Dear Guru?s,

Hi, I have now built a Suse 8.2 box for *

My AVM C2 ISDN card is recognised with no problem. I can call between
extensions, leave voicemail and retrieve same voicemail.

However,

I cannot configure my AVM C2 to either receive calls or make calls ?
HELP!!!!

I am in the UK if that helps or gives a pointer about our ISDN?

Please find below my config: -

; CAPI config
;
[general]
mode=immediate
isdnmode=multipoint
;nationalprefix=0
;internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

msn=01723343950
incomingmsn=01723343950
controller=1
;softdtmf=1
;accountcode;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343951
incomingmsn=01723343951
controller=1
;softdtmf=1
;accountcode;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343952
incomingmsn=01723343952
controller=1
;softdtmf=1
;accountcode;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343953
incomingmsn=01723343953
controller=1
;softdtmf=1
;accountcode;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343954
incomingmsn=01723343954
controller=1
;softdtmf=1
;accountcode;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343955
incomingmsn=01723343955
controller=1
;softdtmf=1
;accountcode;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343956
incomingmsn=01723343956
controller=1
;softdtmf=1
;accountcode;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343957
incomingmsn=01723343957
controller=1
;softdtmf=1
;accountcode;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343958
incomingmsn=01723343958
controller=1
;softdtmf=1
;accountcode;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343959
incomingmsn=01723343959
controller=1
;softdtmf=1
;accountcode;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

;PointToPoint (55512-0)
;for outgoing calls use example 5551212
;and in dialplan you can use callerid like
;exten => _0XXX.,1,StripMSD,1
;exten => _XXX.,2,Dial,CAPI/55512${CALLERIDNUM}:bBYEXTENSION
;===========================================================;mode=immediate
;isdnmode=ptp
;msn=55512
;controller=2
;devices=30


extensions.conf
;
; Static extension configuration files, used by
; the pbx_config module.
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

; You can include other config files, use the #include command (without the
';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk configuration
files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be
referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp				; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
;IAXINFO=guest					; IAXtel username/password
;IAXINFO=myuser:mypass
;TRUNK=Zap/g2					; Trunk interface
;TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider
TRUNK=CAPI
[trunklocal]
exten => _01723.,1,Dial(CAPI/@343955:${EXTEN}|30|r)
[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1			; Wait a second, just for fun
exten => s,2,Answer			; Answer the line
exten => s,3,DigitTimeout,5		; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10		; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats)	; Play a congratulatory message
exten => s,6,BackGround(demo-instruct)	; Play some instructions

exten => 2,1,BackGround(demo-moreinfo)	; Give some more information.
exten => 2,2,Goto(s,6)

exten => 3,1,SetLanguage(fr)		; Set language to french
exten => 3,2,Goto(s,5)			; Start with the congratulations

exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip)		; "Please hold while..."
					; (but skip if channel is not up)
exten => 1234,2,Macro(stdexten,1234,${CONSOLE})

exten => 1235,1,Voicemail(u1234)		; Right to voicemail

exten => 1236,1,Dial(Console/dsp)		; Ring forever
exten => 1236,2,Voicemail(u1234)		; Unless busy

;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks)		; "Thanks for trying the demo"
exten => #,2,Hangup			; Hang them up.

[sip]
exten => 5521,1,Dial(SIP/5521,20,tr)
exten => 5521,2,VoiceMail,u5521
exten => 5521,102,VoiceMail,b5521
exten => 6521,1,Ringing
exten => 6521,2,Wait(2)
exten => 6521,3,VoiceMailMain,s5521

exten => 5526,1,Dial(SIP/5526,20,tr)
exten => 5526,2,VoiceMail,u5526
exten => 5526,102,VoiceMail,b5526
exten => 6526,1,Ringing
exten => 6526,2,Wait(2)
exten => 6526,3,VoiceMailMain,s5526


Your help would be very very welcome as I think I am going out of my mind!!!

Many thanks for all your help.

Regards

Nick

From:		Nick Grindley
Position:	Managing Director / CEO
Company:	Intelligent Television and Video Limited
Country:	United Kingdom
E-Mail:		mailto:npg@itvv.co.uk
www:		www.itvv.co.uk
Tel:		+44 1723 500767 - Extension 527
Fax:		+44 1723 501208

This e-mail, and any attachment, is confidential. If you have received it in
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Television and Video Limited, unless specifically stated.

Hi Nick,

Unless I mistaken you don't seem to have a default context in
extensions.conf so if the CAPI side of things is working it 
will have no where to send the calls. Try renaming the demo
context to 'default', or change the CAPI config to deliver
calls to the 'demo' context

Also, what does /usr/sbin/asterisk -vvvvvc report when you
attempt to call in via one of the ISDN MSNs ?

-Nathan

Nick,

you give us no info what you have tested so far and what are the results. I 
guess you have installed chan_capi.

Two things. May be you have checked that
if you start asterisk so you see anything about CAPI?

what do you get in the asterisk CLI when you enter: capi info?

If all this is ok I would make the following change in extensions.conf

( I I had to do it this way in my extensions.conf, even the context name in 
capi.conf was different. I do not understand why but it works)

hope this helps

Jakob

[default]
include -> demo
>[demo]
>;
>; We start with what to do when a call first comes in.
>;
>exten => s,1,Wait,1                     ; Wait a second, just for fun
>exten => s,2,Answer                     ; Answer the line
>exten => s,3,DigitTimeout,5             ; Set Digit Timeout to 5 seconds
>exten => s,4,ResponseTimeout,10         ; Set Response Timeout to 10
seconds
>exten => s,5,BackGround(demo-congrats)  ; Play a congratulatory message
>exten => s,6,BackGround(demo-instruct)  ; Play some instructions
<snip>