asterisk users - Jun 2005 - unamble to dialout to mobiles and others "special" numbers

Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1
The system is connected with an HFC card directly to the telco line
card is in TE mode
and signalling used is bri_cpe_ptmp

I am able to dial out some "numbers" and some not.
In particular it seems that i can't call mobiles and special telco 
numbers like the information call center, emergency numbers,...

If i use a normal hardware isdn phone i am able to do such calls.

This is a call that "works":

     -- Executing NoOp("SIP/11-1ecc", "Call to 756756756")
in new stack
     -- Executing GotoIf("SIP/11-1ecc", "0?3:5") in new
stack
     -- Goto (default,059305698,5)
     -- Executing GotoIf("SIP/11-1ecc", "0?6:8") in new
stack
     -- Goto (default,059305698,8)
     -- Executing NoOp("SIP/11-1ecc", "External call") in
new stack
     -- Executing Goto("SIP/11-1ecc", "esterni|756756756|1")
in new stack
     -- Goto (esterni,059305698,1)
     -- Executing Dial("SIP/11-1ecc", "Zap/g1/756756756") in
new stack
     -- Requested transfer capability: 0x00 - SPEECH
     -- Called g1/756756756
     -- Zap/1-1 is ringing
[********************now i hangup]
     -- Hungup 'Zap/1-1'
   == Spawn extension (esterni, 756756756, 1) exited non-zero on 
'SIP/11-1ecc'
     -- Executing Goto("SIP/11-1ecc", "default|h|1") in new
stack
     -- Goto (default,h,1)
     -- Executing Hangup("SIP/11-1ecc", "") in new stack
   == Spawn extension (default, h, 1) exited non-zero on 'SIP/11-1ecc'
   == Primary D-Channel on span 1 down
   == Primary D-Channel on span 1 up


This is a call that does NOT work (ir. i'm calling my mobile phone):

   == Primary D-Channel on span 1 down
   == Primary D-Channel on span 1 up
     -- Executing NoOp("SIP/11-9d74", "Call to 37777777") in
new stack
     -- Executing GotoIf("SIP/11-9d74", "0?3:5") in new
stack
     -- Goto (default,37777777,5)
     -- Executing GotoIf("SIP/11-9d74", "0?6:8") in new
stack
     -- Goto (default,3473042866,8)
     -- Executing NoOp("SIP/11-9d74", "External call") in
new stack
     -- Executing Goto("SIP/11-9d74", "esterni|37777777|1")
in new stack
     -- Goto (esterni,37777777,1)
     -- Executing Dial("SIP/11-9d74", "Zap/g1/37777777") in
new stack
     -- Requested transfer capability: 0x00 - SPEECH
     -- Called g1/37777777
     -- Channel 0/1, span 1 got hangup
Jun 16 13:07:17 WARNING[17330]: app_dial.c:412 wait_for_answer: Unable 
to forward voice
Jun 16 13:07:17 WARNING[17330]: app_dial.c:412 wait_for_answer: Unable 
to forward voice
     -- Hungup 'Zap/1-1'
   == No one is available to answer at this time
     -- Executing Answer("SIP/11-9d74", "") in new stack
     -- Executing Playtones("SIP/11-9d74", "congestion") in
new stack
     -- Executing Congestion("SIP/11-9d74", "") in new stack



Some configuration files:
http://marcopar.altervista.org/extensions.conf
http://marcopar.altervista.org/zapata.conf
http://marcopar.altervista.org/zaptel.conf

in the system messages i'm getting this:

Zapata Telephony Interface Registered on major 196
PCI: Enabling device 0000:00:06.0 (0000 -> 0003)
ACPI: PCI interrupt 0000:00:06.0[A] -> GSI 17 (level, low) -> IRQ 185
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd08eaf00 fifo 
0xcf338000(0xf338000) IRQ 185 HZ 1000
zaphfc: Card 0 configured for TE mode
zaphfc: 1 hfc-pci card(s) in this box.

Zaptel Configuration
=====================
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)

Channel map:

Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)

3 channels configured.


frequently i get:
zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, 
card = 0).

Hi,

> I am able to dial out some "numbers" and some not.
> In particular it seems that i can't call mobiles and special telco 
> numbers like the information call center, emergency numbers,...
try with:
pridialplan=unknown
prilocaldialplan=unknown

matteo

-- 
Matteo Brancaleoni
System Administrator
Tel  +39.02.70633354
Sip  matteo@sip.voismart.it
Iax2 matteo@iax.voismart.it