[Asterisk-Users] Dialout from SIP to PSTN

[Andreas Czerniak]

Hi,

i install the Asterisk PBX on a linux machine with i4l to connect to PSTN 
(EuroISDN). And i configure a very simple dial plan in extension.conf.

After this, i connect with a SIP program to asterisk and would call my 
cellular phone, but got this error:

    -- Executing Ringing("SIP/ACzerniak-0904", "") in new stack
    -- Executing Dial("SIP/ACzerniak-0904", "Modem/g1/01xxxxxxxxx") in new 
stack
       chan_modem.c:181 modem_call: Destination g1/01xxxxxxxxx requres a 
real destination (device:destination)
    -- Couldn't call g1/01xxxxxxxxx
    -- Hungup 'Modem[i4l]/ttyI1'
  == Everyone is busy at this time
    -- Executing Congestion("SIP/ACzerniak-0904", "") in new stack
  == Spawn extension (default, 901xxxxxxxxx, 3) exited non-zero on 
'SIP/ACzerniak-0904'

I change the TRUNK variable from Modem/g1 to Modem/ttyI[0|1], but this have 
the same effect.

What means the "Destination g1/01xxxxx requires a _real_ destination" ?

Thanks in advanced.

Regards,
		Andreas.



The modem.conf:

[interfaces]
context=remote
driver=i4l
dialtype=tone
mode=immediate
group=1
msn=85xxxxxx
device => /dev/ttyI0
device => /dev/ttyI1

The extentsion.conf

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
IAXINFO=guest                                   ; IAXtel username/password
TRUNK=Modem/g1
TRUNKMSD=1                                      ; MSD digits to strip 
(usually
PHONE1=SIP/ACzerniak

The [default] section includes:

exten => _90ZXXXXXXXXX,1,Ringing		; read it from
exten => _90ZXXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _90ZXXXXXXXXX,3,Congestion

--
"If you want to pray. Go to the sea."
----------------------------------------------------------------
Andreas Czerniak <cognac at amcs.net>
PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=get&search=0xEDB224EC