Problem und Hilfe gesucht: Kann nicht intern mit ISDN telfonieren !

[astarisk]

Hallo,

habe die Trixbox 2.0 installiert. 2 Cologne-Chip Karten drin. Eine im TE eine NT Modus. Sieht, denke ich, so aus, wie es aussehen muss.

Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" AMI/CCS

1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
Span 2: ZTHFC2 "HFC-S PCI A ISDN card 1 [NT] layer 1 ACTIVATED (G3)" AMI/CCS

4 ZTHFC2/0/1 Clear (In use)
5 ZTHFC2/0/2 Clear (In use)
6 ZTHFC2/0/3 HDLCFCS (In use)

Ich kann auch mit SIP per ISDN (Span 2) nach aussen telefonieren und von aussen Telefonate (ebenfalls SIP, Span 2) entgegen nehmen.

Was nicht klappt, ich kann weder von SIP nach intern ISDN telfonieren:

-- Executing Macro("SIP/200-fd09", "exten-vm|novm|301") in new stack
-- Executing Macro("SIP/200-fd09", "user-callerid") in new stack
-- Executing NoOp("SIP/200-fd09", "user-callerid: device 200") in new stack
-- Executing GotoIf("SIP/200-fd09", "0?report") in new stack
-- Executing GotoIf("SIP/200-fd09", "0?start") in new stack
-- Executing Set("SIP/200-fd09", "REALCALLERIDNUM=200") in new stack
-- Executing NoOp("SIP/200-fd09", "REALCALLERIDNUM is 200") in new stack
-- Executing Set("SIP/200-fd09", "AMPUSER=200") in new stack
-- Executing Set("SIP/200-fd09", "AMPUSERCIDNAME=XXXXXXXXX") in new stac
k
-- Executing GotoIf("SIP/200-fd09", "0?report") in new stack
-- Executing Set("SIP/200-fd09", "CALLERID(all)=Roland Balazs <200>") in new
stack
-- Executing Set("SIP/200-fd09", "REALCALLERIDNUM=200") in new stack
-- Executing NoOp("SIP/200-fd09", "TTL: ARG1: novm") in new stack
-- Executing GotoIf("SIP/200-fd09", "0?continue") in new stack
-- Executing Set("SIP/200-fd09", "_TTL=64") in new stack
-- Executing GotoIf("SIP/200-fd09", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/200-fd09", "Using CallerID "XXXXXXXXXXXX" <200>") in
new stack
-- Executing Set("SIP/200-fd09", "FROMCONTEXT=exten-vm") in new stack
-- Executing Set("SIP/200-fd09", "VMBOX=novm") in new stack
-- Executing Set("SIP/200-fd09", "EXTTOCALL=301") in new stack
-- Executing Set("SIP/200-fd09", "CFUEXT=") in new stack
-- Executing Set("SIP/200-fd09", "CFBEXT=") in new stack
-- Executing Set("SIP/200-fd09", "RT=") in new stack
-- Executing Macro("SIP/200-fd09", "record-enable|301|IN") in new stack
-- Executing GotoIf("SIP/200-fd09", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/200-fd09", "recordingcheck|20070311-151945|asteris
k-4639-1173622784.195") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20070311-151945|asterisk-4639-1173622784.195: Inbound recording
not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/200-fd09", "No recording needed") in new stack
-- Executing Macro("SIP/200-fd09", "dial||tr|301") in new stack
-- Executing DeadAGI("SIP/200-fd09", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: priority is 1
dialparties.agi: Caller ID name is 'XXXXXXXXXXXXX' number is '200'
dialparties.agi: Methodology of ring is 'none'
> dialparties.agi: USE_CONFIRMATION: 'FALSE'
> dialparties.agi: RINGGROUP_INDEX: ''
-- dialparties.agi: Added extension 301 to extension map
-- dialparties.agi: Extension 301 cf is disabled
-- dialparties.agi: Extension 301 do not disturb is disabled
> dialparties.agi: extnum: 301
> dialparties.agi: exthascw: 1
> dialparties.agi: exthascfb: 0
> dialparties.agi: extcfb:
> dialparties.agi: exthascfu: 0
> dialparties.agi: extcfu:
-- dialparties.agi: dbset CALLTRACE/301 to 200
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/200-fd09", "ZAP/1-2||tr") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called 1-2
-- Channel 0/1, span 1 got hangup, cause 28
-- Channel 0/1, span 1 received AOC-E charging 0 units
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing GosubIf("SIP/200-fd09", "0?docfu|1") in new stack
-- Executing GosubIf("SIP/200-fd09", "0?docfb|1") in new stack
-- Executing NoOp("SIP/200-fd09", "Voicemail is novm") in new stack
-- Executing GotoIf("SIP/200-fd09", "1?s-CHANUNAVAIL|1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing PlayTones("SIP/200-fd09", "congestion") in new stack
-- Executing Congestion("SIP/200-fd09", "10") in new stack
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/
200-fd09' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/
200-fd09'

Noch kann ich von intern ISDN nach SIP oder extern ISDN telefonieren. Asterisk erkennt, zwar dass sich was tut, hängt dann aber wieder auf.
Offensichtlich falsch ist, dass Span 2 angezeigt wird.

-- Accepting voice call from '307533' to 's' on channel 0/2, span 2
-- Executing Macro("Zap/5-1", "hangupcall") in new stack
-- Executing ResetCDR("Zap/5-1", "w") in new stack
-- Executing NoCDR("Zap/5-1", "") in new stack
-- Executing GotoIf("Zap/5-1", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("Zap/5-1", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("Zap/5-1", "5") in new stack
-- Executing Hangup("Zap/5-1", "") in new stack
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'Zap/5-1' in
macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'Zap/5-1'
-- Hungup 'Zap/5-1'

Im Anhang habe ich meine gesamten zapata*.conf Dateien angehängt. Ich vermute, dass der Fehler dort begraben liegt. Wie Ihr seht, habe ich schon mit dem Anlegen von Extensions gespielt, um verschiedene Sache auszuprobieren. Aber ich komme so nicht weiter.

Hoffentlich könnt Ihr mit helfen. Wenn Ihr dazu weitere conf-Dateien oder ähnliches benötigt, dann bitte einfach melden.

Danke

Roland

 

[astarisk]

Thema geschlossen. Bin einen Schritt weiter. Neues Problem neuer Thread.

Danke

Roland