Busy/congestion

Hello,
thank you for your answer!

But I have a problem. When I try to phone to ISDN:

-- Executing [999@phones] Verbose("SIP/700-082abde8", "1|-phones- Da "700" <700> a 999 Chiamata verso ISDN") in new stack
1|-phones- Da 700 <700> a 999 Chiamata verso ISDN
-- Executing [999@phones] Dial("SIP/700-082abde8", "mISDN/4/999") in new stack
P[ 4] channel with stid:0 for one second still in use!
P[ 0] --> * NEW CHANNEL dad:999 oad:(null)
P[ 4] * Queuing chan 0x82b1dd0
P[ 4] read_config: Getting Config
P[ 4] --> TON: Unknown
P[ 4] --> LTON: Unknown
P[ 4] --> CTON: Unknown
P[ 4] * CALL: 4/999
P[ 4] --> * dad:999 tech:mISDN/6-u0 ctx:phones
P[ 4] --> * adding2newbc ext 999
P[ 4] --> * adding2newbc callerid 700
P[ 4] --> pres: -1 screen: -1
P[ 4] --> pres: 0
P[ 4] --> PRES: Allowed (0x0)
P[ 4] --> SCREEN: Unscreened (0x0)
P[ 4] NO OPTS GIVEN
P[ 4] I SEND:SETUP oad:700 dad:999 pid:2
P[ 4] --> bc_state:BCHAN_CLEANED
P[ 4] --> channel:0 mode:TE cause:16 ocause:16 rad: cad:
P[ 4] --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0
P[ 4] --> caps:Speech pi:0 keypad: sending_complete:0
P[ 4] --> screen:0 --> pres:0
P[ 4] --> addr:0 l3id:0 b_stid:0 layer_id:0
P[ 4] --> facility:Fac_None out_facility:Fac_None
P[ 4] --> found chan: 1
P[ 4] set_chan_in_stack: 1
P[ 4] --> found channel: 1
P[ 4] --> new_l3id 90001
P[ 4] --> * SEND: State Dialing pid:2
P[ 4] Sending msg, prim:30580 addr:41000404 dinfo:90001
-- Called 4/999
P[ 0] MGMT: SSTATUS: L1_DEACTIVATED
P[ 4] handle_frm: frm->addr:42000403 frm->prim:3f182
P[ 4] --> lib: RELEASE_CR Ind with l3id:90001
P[ 4] --> lib: CLEANING UP l3id: 90001
P[ 4] --> queue_hangup
P[ 4] * RELEASING CHANNEL pid:2 ctx:phones dad:999 oad:999 state: CALLING
P[ 4] --> * State Down
P[ 4] --> Setting AST State to down
P[ 4] $$$ CLEANUP CALLED pid:2
P[ 4] empty_chan_in_stack: 1
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [999@phones] Hangup("SIP/700-082abde8", "") in new stack
== Spawn extension (phones, 999, 3) exited non-zero on 'SIP/700-082abde8'

PS:
The red lights at the back are constantly blinking even when the cable is plugged in.

Hi,

go to the asterisk CLI (asterisk -r) and post the output of the "misdn show stacks" command. There is a number of reasons why you obtain a congestion tone. Anyway your bri connections are down, in fact the leds on the card blink only when connection is down on layer 1/layer 2.

OK,

*CLI> misdn show stacks
BEGIN STACK_LIST:
* Port 1 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:4
* Port 2 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:4
* Port 3 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:4
* Port 4 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:4

I have similar problem
running ubuntu 8.04 hardy amd64 server
asterisk 1.4.22
my ligths blink on b400p
when cable i connected but
when i call they go green
the when i hang up they red but not blinking
and then start to blink again.
Anyway i got diffrent problem
when i call from pstn to sip phone i do not here
the calling party but the party can here me.
I thing the problem is with amd64 but maybe i am wrong
could you send us you asterisk/misdn.conf so i can compare it
to my

Yes of course , but I haven't AMD64

[general]

; set debugging flag:
; 0 - No Debug
; 1 - mISDN Messages and * - Messages, and * - State changes
; 2 - Messages + Message specific Informations (e.g. bearer capability)
; 3 - very Verbose, the above + lots of Driver specific infos
; 4 - even more Verbose than 3
;
debug=4

; the big trace
;tracefile=/var/log/misdn.trace

;
; single call trace files
; set to true if you want to have them
; they depend on debug level
;
trace_calls=true
trace_dir=/var/log

bridging=no

; users sections:
; name your sections as you which but not "general" !
; the secions are Groups, you can dial out in extensions.conf
; with Dial(mISDN/g:extern/101) where extern is a section name,
; chan_misdn tries every port in this section to find a
; new free channel


;
; The default section is not a group section, it just contains config elements
; which are inherited by group sections.
;
[default]
echocancel=1000
echotraining=500

[incall]
; Solo chiamate da ISDN a Asterisk
ports=1,2,3
context=phones

[outcall]
; Solo chiamate da Telefono a ISDN
ports=4
context=phones

Intersting thing
i use isdn bri for call from pstn (telephone company)
and i have
[inext]
always_immediate=yes
ports=1
context=ulaz1
bridging=no


the context is maped to sip phone