Outgoing calls with Eicon Diva Server 4BRI


#1

Hi!
Can anyone help me with outgoing calls please?
I’m using Asterisk on Linux/Ubuntu with a
Eicon Diva Server 4BRI
(http://www.eicon.com/us/products/MediaGateways/DIVA_4BRI_8M.htm)
I have this problem: if I create a spefic extension
in extensions.conf like this:

(it works)
exten => 123,1,Dial(CAPI/ISDN1:phone_number)

(it doesn’t work)
exten => _9.,1,Dial(CAPI/ISDN1:{exten:1})
Dialing this entry, I receive the message: 403: forbidden

Can anyone help me please?!?

(My boss is asking me every day about it, :cry:
he want me to configure it as soon as possible)

Excuse me for my bad english, sorry! :wink:

Giuseppe


#2

If you’re using chan_capi-cm, you should use the following syntax:

Dial(CAPI/ISDN1/{exten:1}/b)


#3

[quote=“Djelibeybi”]If you’re using chan_capi-cm, you should use the following syntax:

Dial(CAPI/ISDN1/{exten:1}/b)[/quote]

I tryed to use your line:
Dial(CAPI/ISDN1/{exten:1}/b)
but I get again the error: 403 forbidden

Should I set something about groups?
Here there are my settings… so meybe someone can
tell me where is the problem! Help me please! Tanks a lot!

Giuseppe

This is my sip.conf
--------------------BEGIN--------------------
[general]
context=default
bindport=5060 ; 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=it ; Default language setting for all users/peers

[giuseppe]
;Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
secret=xxxxxxxx
host=dynamic
nat=no
callerid=Giuseppe102 ;definisce la stringa inviata dal chiamante al destinatario al posto di quella di default
canreinvite=no
context=internal
disallow=all
callgroup=1 ; We are in caller groups 1,3,4
pickupgroup=1 ; We can do call pick-p for call group 1,3,4,5
allow=gsm
allow=ulaw
allow=alaw

[luca]
;Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
secret=xxxxxxxx
host=dynamic
nat=no
callerid=luca101 ;definisce la stringa inviata dal chiamante al destinatario al posto di quella di default
canreinvite=no
context=internal
disallow=all
callgroup=1 ; We are in caller groups 1,3,4
pickupgroup=1 ; We can do call pick-p for call group 1,3,4,5
allow=gsm
allow=ulaw
allow=alaw

--------------------END--------------------

This is my capi.conf
--------------------BEGIN--------------------

[general]
nationalprefix=039
;;internationalprefix=011
rxgain=0.8
txgain=0.8
;rxgain=1.0
;txgain=1.0
;;ulaw=yes ;set this, if you live in u-law world instead of a-law

; interface sections …

[ISDN1] ;this example interface gets name ‘ISDN1’ and may be any
ntmode=yes ;if isdn card operates in nt mode, set this to yes when using NT-mode, ptp should be set in any case
incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * == any
controller=1 ;capi controller number to use
group=1 ;dialout group
;softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards
;relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection
;accountcode= ;Asterisk accountcode to use in CDRs
context=isdn1in ;context for incoming calls
holdtype=local ;If set to ‘hold’, when Asterisk puts the call on hold, ISDN HOLD will be used.
;If set to ‘local’ (default value), no hold is done and Asterisk may play MOH.
immediate=yes ;immediate start of pbx with extension ‘s’ if no digits were received on incoming call (no destination number yet)
;echosquelch=1 ;VERY_PRIMITIVE echo suppression
;; ----- l’echocancel se abilitato non funziona ancora, genera un errore in asterisk -v (controllare)
;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation (possible values: ‘no’, ‘yes’, ‘force’, ‘g164’, ‘g165’)
;echocancelold=yes ;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)
echotail=64 ;echo cancel tail setting
bridge=yes ;native bridging (CAPI line interconnect) if available
callgroup=1 ;Asterisk call group
;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are busy
devices=2 ;number of concurrent calls on this controller (2 makes sense for single BRI, 30 for PRI)

-------------------END---------------------

This is my capi.conf
--------------------BEGIN--------------------
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=yes ;default: NO

[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)

[internal]
exten => 101,1,Ringing ; simula che il tel remoto stia squillando fino alla successiva istruzione
exten => 101,2,Wait,3 ; mette in attesa il chiamante per N secondi (Wait,N)
exten => 101,3,Dial(Sip/lucao,60,rt) ; dial(destinatario,timeout,parametro [vedi pag 243,244 manuale asterisk o’reilly])
exten => 102,1,Ringing ; simula che il tel remoto stia squillando fino alla successiva istruzione
exten => 102,2,Wait,3 ; mette in attesa il chiamante per N secondi (Wait,N)
exten => 102,3,Dial(Sip/giuseppedd,60,rt) ; dial(destinatario,timeout,parametro [vedi pag 243,244 manuale asterisk o’reilly])
exten => _9.,1,Dial(CAPI/ISDN1/{exten:1}/b)
-------------------END---------------------