[email protected] 2xHFC an Teles PBX raustelefonieren geht nicht

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Hallo,
bin "neu" hier und hab folgendes Problem:

[email protected] mit Dondisperato- Anleitung für HFC gepatched
2 x HFC (benutze erst eine! = 2. nicht konfiguriert/angeschlossen)
2 SIP Telefone
Anschluß an Teles DOS PABX V292t über internen Anschluß

1. HFC konfiguriert auf TE ( PABX hat ja NT = Kabel 1:1! )

Hab alles konfiguriert nach etlichen Fehlversuchen ist derzeitiger Stand:
SIP <==> SIP geht
SIP ==> extern über SIP geht
extern ==> SIP geht
extern ==> PABX ==> SIP Tel. geht
SIP Tel. ==> extern über PABX -------> all lines are Busy now

Ich kann in den Threads hier leider nichts passendes finden...

Irgendwo gibt es noch einen Fehler beim "raustelefonieren" über die
ISDN- Telefonanlage...
Kann auch keine Verbindung zu internen Telefonen der ISDN.- Anlage herstellen...
Hab im AMP unter ausgehenden Vbs mit der 8 einen Weg über ZAP/g1
eingestellt
Kann mir bitte jemand helfen???!!!

Gruß
Enno

Hier noch ein paar Daten:

Zapata.conf
==========
[channels]

language=de
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=no
context=from-pstn
channel => 1-2

Zapata_additional.conf:
==================
;;;;;;[32]
signalling=bri_cpe_ptmp
record_out=Adhoc
record_in=Adhoc
mailbox=32@device
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callerid=device <32>
busydetect=no
busycount=7
channel=>g2

extensions.conf ( Auszüge )
======================
[from-pstn]
exten => 24,1,NoOp(Incoming call on ISDN Span 1 forSIP 24)
exten => 24,2,Dial(local/24@from-internal,20,m)
exten => 24,3,VoiceMail(24@default)
exten => 24,4,Hangup
exten => 33,1,NoOp(Incoming call on ISDN Span 1 for SIP 33)
exten => 33,2,Dial(local/33@from-internal,20,m)
exten => 33,3,VoiceMail(33@default)
exten => 33,4,Hangup
exten => s,1,NoOp(Incoming call on ISDN unknown - Signaling all = SIP 24 + 33)
exten => s,2,Dial(local/24&local/33@from-internal,20,m)
exten => s,3,VoiceMail(24@default)
exten => s,4,Hangup


include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations
include => ext-did
include => from-pstn-timecheck ; this has to be included otherwise it overrides ext-did
exten => fax,1,Goto(ext-fax,in_fax,1)

------------------------------------------

Hier ein Debug von einem Call:
=======================
-- Executing Macro("SIP/24-d8ff", "dialout-trunk|4|0061817090712|") in new stack
-- Executing GotoIf("SIP/24-d8ff", "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/24-d8ff", "user-callerid") in new stack
-- Executing DBget("SIP/24-d8ff", "AMPUSER=DEVICE/24/user") in new stack
-- DBget: varname=AMPUSER, family=DEVICE, key=24/user
-- DBget: set variable AMPUSER to 24
-- Executing DBget("SIP/24-d8ff", "AMPUSERCIDNAME=AMPUSER/24/cidname") in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=24/cidname
-- DBget: set variable AMPUSERCIDNAME to SBR24_Halle
-- Executing GotoIf("SIP/24-d8ff", "0?5") in new stack
-- Executing SetCallerID("SIP/24-d8ff", ""SBR24_Halle" <24>") in new stack
-- Executing NoOp("SIP/24-d8ff", "Using CallerID "SBR24_Halle" <24>") in new stack
-- Executing Macro("SIP/24-d8ff", "record-enable|24|OUT") in new stack
-- Executing GotoIf("SIP/24-d8ff", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/24-d8ff", "recordingcheck|20060413-143820|asterisk-3581-1144931900.18") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060413-143820|asterisk-3581-1144931900.18: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/24-d8ff", "No recording needed") in new stack
-- Executing Macro("SIP/24-d8ff", "outbound-callerid|4") in new stack
-- Executing DBget("SIP/24-d8ff", "USEROUTCID=AMPUSER/24/outboundcid") in new stack
-- DBget: varname=USEROUTCID, family=AMPUSER, key=24/outboundcid
-- DBget: set variable USEROUTCID to
-- Executing GotoIf("SIP/24-d8ff", "1?4") in new stack
-- Goto (macro-outbound-callerid,s,4)
-- Executing GotoIf("SIP/24-d8ff", "1?6") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing NoOp("SIP/24-d8ff", "CallerID set to "SBR24_Halle" <24>") in new stack
-- Executing SetGroup("SIP/24-d8ff", "OUT_4") in new stack
-- Executing CheckGroup("SIP/24-d8ff", "2") in new stack
-- Executing SetVar("SIP/24-d8ff", "DIAL_NUMBER=0061817090712") in new stack
-- Executing SetVar("SIP/24-d8ff", "DIAL_TRUNK=4") in new stack
-- Executing AGI("SIP/24-d8ff", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("SIP/24-d8ff", "OUTNUM=0061817090712") in new stack
-- Executing Cut("SIP/24-d8ff", "custom=OUT_4|:|1") in new stack
-- Executing GotoIf("SIP/24-d8ff", "0?16") in new stack
-- Executing Dial("SIP/24-d8ff", "ZAP/g1/0061817090712") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/24-d8ff", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/24-d8ff", "Dial failed due to CHANUNAVAIL") in new stack
-- Executing Macro("SIP/24-d8ff", "outisbusy") in new stack
-- Executing Playback("SIP/24-d8ff", "all-circuits-busy-now") in new stack
-- Playing 'all-circuits-busy-now' (language 'de')
-- Executing Playback("SIP/24-d8ff", "pls-try-call-later") in new stack
-- Playing 'pls-try-call-later' (language 'de')
-- Executing Macro("SIP/24-d8ff", "hangupcall") in new stack
-- Executing ResetCDR("SIP/24-d8ff", "w") in new stack
-- Executing NoCDR("SIP/24-d8ff", "") in new stack
-- Executing Wait("SIP/24-d8ff", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/24-d8ff' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/24-d8ff' in macro 'outisbusy'
== Spawn extension (from-internal, 80061817090712, 2) exited non-zero on 'SIP/24-d8ff'
-- Executing Macro("SIP/24-d8ff", "hangupcall") in new stack
-- Executing ResetCDR("SIP/24-d8ff", "w") in new stack
-- Executing NoCDR("SIP/24-d8ff", "") in new stack
-- Executing Wait("SIP/24-d8ff", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/24-d8ff' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/24-d8ff'


-----------------------------------------------

So, hab ich noch was vergessen? - Ach JA --- HILFE !
 
Hi,

bin inzwischen etwas weiter...

Es liegt wohl an dem nicht existenten Channel g1 ...
Ersetzt man beim DIAL/g1 durch 1 dann bekomme ich wenigstens verbindung in die Telefonanlage und kann jetzt munter zwischen den SIP und ISDN Tel.
verbinden...
 
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