Asterisk am PMX anschluss

[asterisksip]

Hallo, wir haben asterisknow auf einem System zulaufen über ein Pmx 2Mbit anschluus. Jetzt habe ich einen neuen Server auf dem ich asterisk Asterisk 1.4.24 installiert auf centos5.2.


Die Einstellungen habe ich soweit es geht wie von dem asterisknow system übernommen und ich konnte EINgeschrängt Telefonieren.
Nur wenn ich die Vorwahl ohne Null eingeb kann ich raustelefonieren.
Wenn ich nur die Festnetznr. ohne Vorwahl einegebe geht es garnicht.
Bei Handys muss ich ebenfalls die NUll weg lassen.
Mittlerweile klappt garnichts mehr :-(

hier meine Konfig Dateien:

/etc/zaptel.conf

# Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER) HDB3/CCS
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16

# Global data

loadzone        = us
defaultzone     = us           de hatte ich auch schon drin

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

/etc/asterisk/zapata.conf

File zapata.conf.zapscan not changed so no update needed
[root@voip1 asterisk]# /etc/init.d/asterisk restart
Shutting down asterisk: Asterisk ended with exit status 0
Asterisk shutdown normally.
                                                           [  OK  ]
Starting asterisk:                                         [  OK  ]
[root@voip1 asterisk]# less /etc/asterisk/zapata.conf
context = default
switchtype = euroisdn
internationalprefix = 00
nationalprefix = 0
localprefix = 030
signalling = pri_cpe
rxwink = 300
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
rxgain = 0.0
txgain = 0.0
group = 1
callgroup = 1
pickupgroup = 1
immediate = no

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

Auszug aus /etc/asterisk/extensions.conf

[globals]
CONSOLE = Console/dsp  ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO = guest  ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK = Zap/G2  ; Trunk interface
;
; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
; in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel
;    (aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy Zap channel
;    (aka. descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than last
;    time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than last
;    time (aka. descending rotary hunt group).
;
TRUNKMSD = 1  ; MSD digits to strip (usually 1 or 0)
FEATURES =
DIALOPTIONS =
RINGTIME = 20
FOLLOWMEOPTIONS =
PAGING_HEADER = Intercom
span_1 = Zap/g1
GLOBAL_OUTBOUNDCID =
GLOBAL_OUTBOUNDCIDNAME =
CID_924 = 30920111112
CID_956 = 30920111111
;TRUNK=IAX2/user:pass@provider


....

[DID_span_1]
include = DID_span_1_timeinterval_tagsueber,${timeinterval_tagsueber}
include = DID_span_1_default

[DID_span_1_default]
exten = 030211111,1,Goto(default,924,1)
exten = 20111111,1,Goto(default,924,1)
exten = 20111112,1,Goto(default,956,1)

[voicemenu-custom-1]
exten = s,1,NoOp(voice-test)
exten = s,2,Playback()

[CallingRule_outgoingnormal]
exten = _.,1,Macro(trunkdial-failover-0.3,${span_1}/${EXTEN:0},,span_1,span_1)

[DLPN_DialPlan1]
include = CallingRule_outgoingnormal
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension

Hatz jemand ne idee wo es hagt mitleerwiele klappen nochnichtmal mehr interne anrufe :-(

Wir verwenden Open Stage Sip Telefone die sich bei Asterisk registrieren.