Newsgroup: it.tlc.telefonia.voip
Titolo: asterisk isdn


Jan  3 17:50:07 VERBOSE[2992] logger.c:     -- Executing 
Dial("SIP/100-2c89", "Zap/1/130") in new stack
Jan  3 17:50:07 NOTICE[2992] app_dial.c: Unable to create channel of type 
'Zap' (cause 0 - Unknown)
Jan  3 17:50:07 VERBOSE[2992] logger.c:   == Everyone is busy/congested at 
this time (1:0/0/1)
Jan  3 17:50:07 DEBUG[2992] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.
Jan  3 17:50:09 DEBUG[2391] chan_sip.c: Stopping retransmission on 
'
02550fe349060f12...@192.0.152.152
' of Request 102: Match Found
Jan  3 17:50:17 WARNING[2992] pbx.c: Timeout, but no rule 't' in context 
'testtest'
Jan  3 17:50:17 DEBUG[2992] cdr_addon_mysql.c: cdr_mysql: inserting a CDR 
record.
Jan  3 17:50:17 DEBUG[2992] cdr_addon_mysql.c: cdr_mysql: SQL command as 
follows: INSERT INTO cdr 
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) 
VALUES ('2006-01-03 17:50:07','\"100\" <100>','100','9130','testtest', 
'SIP/100-2c89','','Dial','Zap/1/130',10,0,'NO 
ANSWER',3,'','asterisk-2346-1136307007.25')
Jan  3 17:50:17 DEBUG[2992] chan_sip.c: update_call_counter(100) - decrement 
call limit counter
Jan  3 17:50:17 DEBUG[2391] chan_sip.c: Stopping retransmission on 
'
D7805E4F-2394-402A...@192.0.0.202
' of Response 64912: Match 
Found
dalla console di asterisk ricevo questo:
$asterisk -r
cli>
Executing Dial("SIP/100-7c7f", "Zap/1/130") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called 1/130
    -- Channel 0/1, span 1 got hangup, cause 42
    -- Zap/1-1 is circuit-busy
    -- Hungup 'Zap/1-1'
OPPURE
-- Executing Dial("SIP/100-2c89", "Zap/1/130") in new stack
  == Everyone is busy/congested at this time (1:0/0/1)
/etc/zaptel.conf
 hfc-s pci a span definition
# most of the values should be bogus because we are not really zaptel
loadzone=it
defaultzone=it
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
-----____________________________________
/etc/asterisk/zapata.conf
[channels]
language=it
switchtype=euroisdn
;If you connect to a hicom PBX set your ISDN Numbering Plan Identifier to 
unknown.
pridialplan=local
prilocaldialplan=local
signalling = bri_cpe_ptmp
;signalling = fxs_ks
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
nationalprefix = 0
internationalprefix = 00
faxdetect=incoming
group=0
callgroup=1
pickupgroup=1
immediate=yes
context=testtest
#context=from-pstn
channel => 1-2
:_______________________________________
/etc/asterisk/sip.conf
[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
canreinvite=no
register => 
69xxxxxxxx:mexxx...@sip.vira.it
[vira]
type=friend
username=6969xxxxxx
fromuser=6969xxxxx
secret=xxxxxxxxx
host=
sip.vira.it
context=from-vira
insecure=very
disallow=all
allow=ilbc
allow=gsm
allow=G729
allow=ulaw
allow=alaw
[100]
type=friend
username=100
secret=100
qualify=yes
nat=yes
host=dynamic
context=testtest
[200]
type=friend
username=200
secret=200
qualify=yes
nat=yes
host=dynamic
context=testtest
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
__________________________
/etc/extensions.conf
[ext-test]
exten => 7777,1,Goto(from-pstn,s,1)
exten => 666,1,Goto(ext-fax,in_fax,1)
exten => h,1,Macro(hangupcall)
;echo test
exten => *43,1,Answer
exten => *43,2,Wait(2)
exten => *43,3,Playback(demo-echotest)
exten => *43,4,Echo
exten => *43,5,Playback(demo-echodone)
exten => *43,6,Hangu
[testtest]
exten => 100,1,Dial(Sip/100,,r)
exten => 200,1,Dial(Sip/200,,r)
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,Dial(Sip/100,30)
exten => s,4,Dial(Sip/200,30)
exten => s,5,Hangup()
exten => _9.,1,Dial(Zap/1/${EXTEN:1})
include => default-vira
;exten => s,1,Answer()
;exten => s,2,Playback(hello-world)
;exten => s,3,Hangup()
;exten => s,1,Answer()
;exten => s,2,Echo()
[default-vira]
exten => _0.,1,Dial(SIP/${EXTEN:1}@vira,30,T)
[from-vira]
exten => s,1,Dial(SIP/100,30,tr)
exten => s,2,Dial(SIP/200,30,tr)
ciao e grazie|||| 

 minuti.
Scusa se non mi sono preso la briga di leggere tutto il resto ma ...
Sei sicuro che il Bus S ,ovvero il collegamento tra la NT e il TA, sia fatto
a regola d'arte ?
Per maggiorni informazioni : 
http://www.isdnfaq.info
Ciao
-- 
_Starlight_

 ha scritto nel messaggio 
news:dpirgn$9se$1...@area.cu.mi.it...

GRAZIE DELLA RISPOSTA, MA UN NORMALE CAVO ISDN UTILIZZATO per il 
collegamento di telefoni isdn alla borchia telecom italia va bene giusto?
Nel nostro caso abbiamo usato un cavo di rete a 10 m/bit diritto.
ciao

Ciao ,   PINO !
Hai scritto   il Fri, 6 Jan 2006 12:12:26 +0100:
 P> Nel nostro caso abbiamo usato un cavo di rete a 10 m/bit diritto.
Sempreche' ci siano 4 coppie ; io ho cavi di rete con 2 coppie soltanto ed
in questo caso e' bene tenere a mente che le coppie usate su ethernet non
sono le stesse di quelle usate su ISDN
Io ho avuto anche la sfiga di incappare in una "ciabatta" ISDN con la presa
di uplink sbagliata per cui il TA ed il router singolarmente andavano mentre
collegati assieme no
Cordiali Saluti
                Pietro Bruno Lancerotto
bruno.la...@zadi.vi.it