hallo,
ich kann meine karte inzwischen erfolgreich initilisieren, aber telefonieren ist noch lang ned ;-(
Die ZAP Channels sind da.
Die Karte sauber geladen
dmesg output
aber wenn ich telefonieren will kommt das:
meine zaptel.conf
meine zapata.conf
meine extensions.conf
ich bin am ende von google dem forum und meines ABC's hat einer ne Idee, oder sogar die Lösung ?
Vielen Dank
bye
eazy
ich kann meine karte inzwischen erfolgreich initilisieren, aber telefonieren ist noch lang ned ;-(
Die ZAP Channels sind da.
Code:
*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo inbound-isdn
1 inbound-isdn
2 inbound-isdn
Die Karte sauber geladen
dmesg output
Code:
Zapata Telephony Interface Registered on major 196
zaphfc: no version for "zt_receive" found: kernel tainted.
ACPI: PCI interrupt 0000:02:0d.0[A] -> GSI 17 (level, low) -> IRQ 17
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xf0895000 fifo 0xeed88000(0x2ed88000) IRQ 17 HZ 1000
zaphfc: Card 0 configured for TE mode
zaphfc: 1 hfc-pci card(s) in this box.
Disabled Privacy Extensions on device c04d67c0(lo)
eth0: no IPv6 routers present
Registered tone zone 3 (Netherlands)
aber wenn ich telefonieren will kommt das:
Code:
-- Executing SetCallerID("SIP/eazy-7e52", "505205") in new stack
-- Executing Dial("SIP/eazy-7e52", "Zap/g1d/01794960714|30") in new stack
Aug 29 14:10:21 NOTICE[1655]: app_dial.c:805 dial_exec: Unable to create channel of type 'Zap'
== Everyone is busy/congested at this time
Aug 29 14:10:31 WARNING[1655]: pbx.c:1952 ast_pbx_run: Timeout, but no rule 't' in context 'sip-phones'
-- parse_srv: SRV mapped to host sipgate.de, port 5060
meine zaptel.conf
Code:
asterisk:~# cat /etc/zaptel.conf
# hfc-s pci a span definition
# most of the values should be bogus because we are not really zaptel
loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
alaw=1-3
meine zapata.conf
Code:
asterisk:~# cat /etc/asterisk/zapata.conf
;
; Zapata telephony interface
;
; Configuration file
[channels]
;
; Default language
;
;language=en
;
; Default context
;
;
switchtype = euroisdn
; p2mp TE mode
signalling = bri_cpe_ptmp
; ^^^^^^ this is for zaphfc modes=0
; p2p TE mode
;signalling = bri_cpe
; p2mp NT mode
;signalling = bri_net_ptmp
; ^^^^ this is for zaphfc modes=1
; p2p NT mode
;signalling = bri_net
pridialplan=local
prilocaldialplan=local
nationalprefix = 0
internationalprefix = 00
; trust user provided callerid (clip no screening)?
pritrustusercid = yes
callerid=asreceived
echocancel=yes
immediate=no
group = 1
context=inbound-isdn
channel => 1-2
meine extensions.conf
Code:
asterisk:~# cat /etc/asterisk/extensions.conf
[sipout]
exten => _9.,1,SetCallerId,3071032
exten => _9.,2,Dial(SIP/${EXTEN:1}@sipgate-out,30,trg)
exten => _9.,3,Hangup
[isdnout]
exten => _0.,1,SetCallerID,505205
exten => _0.,2,Dial(Zap/g1d/${EXTEN:1},30)
[inbound-isdn]
exten => 505205,1, Dial(SIP/eazy,10,t)
exten => 505205,2, Dial(SIP/eazy1,10,t)
[from-sip]
;include => sipout
exten => 3071032,1, Dial(SIP/eazy,10,t)
exten => 3071032,2, Dial(SIP/eazy1,10,t)
[sip-phones]
include => sipout
include => isdnout
exten => 15,1,Dial(SIP/eazy,20,tr)
exten => 15,2,Congestion
exten => 15,102,Busy
exten => 16,1,Dial(SIP/eazy1,20,tr)
exten => 16,2,Congestion
exten => 16,102,Busy
exten => 17,1,Dial(SIP/mak,20,tr)
exten => 17,2,Congestion
exten => 17,102,Busy
ich bin am ende von google dem forum und meines ABC's hat einer ne Idee, oder sogar die Lösung ?
Vielen Dank
bye
eazy