NTBA <-> Asterisk <-> Auerswald Commander Basic

[IngoM]

Hallo,

ich möchte meine vorhandene Auerswald Commander Basic and meine Asterisk Anlage verbinden dabei soll folgende Reihenfolge gelten:

NTBA - HFC - ASTERISK - HFC - AUERSWALD

Ich benutze eine Junghanns quadBRI 2 die mit TE / TE / NT /NT konfiguriert ist.

Der erste Teil (NTBA - HFC - ASTERISK) funktioniert einwandfrei.
Beim 2. Teil komme ich nicht üer meinen Gehirnknoten hinweg und begreife den zusammenschluss nicht.

Die Auerswald CB hat folgende Module 4S0 und 8a/b
Den Eingang der UAerswald (S0) habe ich mit Port 3 (NT) der quadBRI verbunden. Hier die Zapata dazu:

; Zapata telephony interface
;
; Configuration file



[channels]
language=de

; Basic ISDN Card TE mode config

; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode)
; signalling = bri_cpe_ptmp
; p2p TE mode (for connecting ISDN lines in point-to-point mode)
; signalling = bri_cpe
; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode)
; signalling = bri_net_ptmp
; p2p NT mode (for connecting an ISDN pbx in point-to-point mode)
; signalling = bri_cpe

resetinterval=never
immediate=no

switchtype=euroisdn
;signalling=bri_cpe_ptmp
;signalling=bri_cpe

pridialplan=dynamic
prilocaldialplan=local

nationalprefix=0
internationalprefix=00

echotraining=100

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
;faxdetect=incoming

[trunkgroups]
; Junghanns 4 port BRI
signalling = bri_cpe
group = 1
context=from-zaptel
; S/T port 1
channel =1-2
channel =4-5

signalling = bri_cpe
context = from-pbx
group = 2
channel =7-8
channel =10-11


;Include genzaptelconf configs
#include zapata-auto.conf


;Include AMP configs
#include zapata_additional.conf



;Include BRI-HFC configs
#include zapata-BRI-HFC.conf

Dazu habe ich in der Auerswald die Nummer 4500 als MSN/DDI eingerichtet, die dann an ein angeschlossenes ISDN Telefon mittels Extern->Teilnehmer Rufnummernverteilung weitergeleitet wird.

Desweiteren habe ich im Asterisk einen ZAP/Trunk (g2) eingerichtet der als Outbound Caller ID 4500 übergibt.

Ein Outbound Route fängt die Extension 4 ab (4|.) und routed diese Anrufe an den ZAP/g2 Trunk

Mit dem IP phone 501 rufe ich dann z.B. 4100 an. Ich dachte, dass dann die Auerswald über die mitgegebene CID 4500 auf ihrem MSN/DDI 4500 anspricht und den Anruf entsprechend weiterleitet.

Das Protokoll zeigt aber, dass der Call überhaupt nicht aus der ASTERISK heruaskommt (all Circuits are busy).

-- Executing Macro("SIP/501-088e1a38", "dialout-trunk|3|100||") in new stack
    -- Executing Set("SIP/501-088e1a38", "DIAL_TRUNK=3") in new stack
    -- Executing Set("SIP/501-088e1a38", "_NODEST=") in new stack
    -- Executing Set("SIP/501-088e1a38", "DIAL_NUMBER=100") in new stack
    -- Executing Set("SIP/501-088e1a38", "ROUTE_PASSWD=") in new stack
    -- Executing Set("SIP/501-088e1a38", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "1?noauth") in new stack
    -- Goto (macro-dialout-trunk,s,8)
    -- Executing Set("SIP/501-088e1a38", "GROUP()=OUT_3") in new stack
    -- Executing Macro("SIP/501-088e1a38", "user-callerid|SKIPTTL") in new stack
    -- Executing NoOp("SIP/501-088e1a38", "user-callerid: device 501") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "0?report") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "0?start") in new stack
    -- Executing Set("SIP/501-088e1a38", "REALCALLERIDNUM=501") in new stack
    -- Executing NoOp("SIP/501-088e1a38", "REALCALLERIDNUM is 501") in new stack
    -- Executing Set("SIP/501-088e1a38", "AMPUSER=501") in new stack
    -- Executing Set("SIP/501-088e1a38", "AMPUSERCIDNAME=Keller Nicole") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "0?report") in new stack
    -- Executing Set("SIP/501-088e1a38", "CALLERID(all)=Keller Nicole <501>") in new stack
    -- Executing Set("SIP/501-088e1a38", "REALCALLERIDNUM=501") in new stack
    -- Executing NoOp("SIP/501-088e1a38", "TTL:  ARG1: SKIPTTL") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,21)
    -- Executing NoOp("SIP/501-088e1a38", "Using CallerID "Keller Nicole" <501>") in new stack
    -- Executing Macro("SIP/501-088e1a38", "record-enable|501|OUT") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing DeadAGI("SIP/501-088e1a38", "recordingcheck|20070406-075736|asterisk-3047-1175839056.4") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20070406-075736|asterisk-3047-1175839056.4: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/501-088e1a38", "No recording needed") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "0?skipoutcid") in new stack
    -- Executing Set("SIP/501-088e1a38", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing Macro("SIP/501-088e1a38", "outbound-callerid|3") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "1?start") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing NoOp("SIP/501-088e1a38", "REALCALLERIDNUM is 501") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,9)
    -- Executing Set("SIP/501-088e1a38", "USEROUTCID=") in new stack
    -- Executing Set("SIP/501-088e1a38", "EMERGENCYCID=") in new stack
    -- Executing Set("SIP/501-088e1a38", "TRUNKOUTCID=4500") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,16)
    -- Executing GotoIf("SIP/501-088e1a38", "0?usercid") in new stack
    -- Executing Set("SIP/501-088e1a38", "CALLERID(all)=4500") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "1?report") in new stack
    -- Goto (macro-outbound-callerid,s,22)
    -- Executing NoOp("SIP/501-088e1a38", "CallerID set to "" <4500>") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,16)
    -- Executing DeadAGI("SIP/501-088e1a38", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing Set("SIP/501-088e1a38", "OUTNUM=100") in new stack
    -- Executing Set("SIP/501-088e1a38", "custom=ZAP/g2") in new stack
    -- Executing GotoIf("SIP/501-088e1a38", "0?customtrunk") in new stack
    -- Executing Dial("SIP/501-088e1a38", "ZAP/g2/100|300|") in new stack
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Goto("SIP/501-088e1a38", "s-CONGESTION|1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing NoOp("SIP/501-088e1a38", "Dial failed due to CONGESTION - failing through to other trunks") in new stack
    -- Executing Macro("SIP/501-088e1a38", "outisbusy|") in new stack
    -- Executing Playback("SIP/501-088e1a38", "all-circuits-busy-now|noanswer") in new stack
    -- Playing 'all-circuits-busy-now' (language 'en')
    -- Executing Playback("SIP/501-088e1a38", "pls-try-call-later|noanswer") in new stack
    -- Playing 'pls-try-call-later' (language 'en')
  == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/501-088e1a38' in macro 'outisbusy'
  == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/501-088e1a38'

Kann mir jemand bitte sagen, was ich hier (wahrscheinlich grundlegend ) falsch mache.

Gruß und Danke im Voraus

 

[IngoM]

hat denn hier keiner eine Idee zu dem Thema ???:shock: