asterisk users - Oct 2003 - E100P setup in Switzerland

Hi all,

I am trying to setup an E100P for use on Swisscom E1-PRI here in 
Switzerland.

Swisscom seems to use Siemens hardware.

Here are my configs (cvs from a few hours ago) :

zaptel.conf

loadzone=fr ; tried "de" but got warning at modprobe
defaultzone=fr
span=1,1,0,cas,hdb3,crc4,yellow ; Seems to be right, as the Siemens 
SDSL-G703 modem switches its LEDs off ... not sure
bchan=1-15,17-31
dchan=16

zapata.conf

...
switchtype = euroisdn
signalling = pri_cpe
group = 1
channel => 1-15,17-31

When I try to dial-out, I get :

NOTICE[33809]: File app_dial.c, Line 502 (dial_exec): Unable to create 
channel of type 'Zap'
   == Everyone is busy at this time
     -- Executing Congestion("SIP/22-8f32", "") in new stack
   == Spawn extension (from-sip, 32423423423, 2) exited non-zero on 
'SIP/22-8f32'

Any idea ??

Thanks

Quoting Marcel Prisi <marcel@virtua.ch>:
> Swisscom seems to use Siemens hardware.
> 
> Here are my configs (cvs from a few hours ago) :
> 
> zaptel.conf
> 
> loadzone=fr ; tried "de" but got warning at modprobe
> defaultzone=fr
> span=1,1,0,cas,hdb3,crc4,yellow ; Seems to be right, as the Siemens 
> bchan=1-15,17-31
> dchan=16
First of all use ccs not cas, 

When starting asterisk do U see that D channel is up
Are B channels restarted ?
When running zttool do you have OK value on your E1
Can u see the status from siemens side ?


regards
m.

I have some news ... after a bit of tweaking, the following seems to 
work with Swisscom :

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

I still have a problem : Incoming calls from PRI work well, but outgoing 
call don't :

I dial from a Grandstream 101

   -- Executing Dial("SIP/22-9129", "Zap/g1/234234234") in
new stack
     -- Called g1/234234234
     -- Zap/1-1 is ringing

Well ... I hear a ringing tone in the Grandstream, but the called number 
doesn't ring at all ...

What can I do ??

Thanks