zaphfc ISDN Karte im TE Modus

[linuxpingu]

Hallo wo liegt der Hund begraben????

Ich habe Asterisk@Home version 2.6 mit Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f and flortz patch nach Anleitung von http://www.vocesuip.com/viewtopic.php?t=1359&highlight= installiert.

zaphfc im TE Modus am SBus zum NT der swisscom

über ZAP/g0
eingehende Anrufe = OK
abgehende Anrufe = gehen nicht meldung all circuits are busy..

[root@asterisk1 ~]# ztcfg -vvv

Zaptel Configuration
======================

SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)

3 channels configured.

zapata.conf
;
; Zapata telephony interface
;
; Configuration file

[channels]
language=it
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

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


extensions_additional.conf

[globals]
#include globals_custom.conf
VM_PREFIX = *
RINGTIMER = 60
REGTIME = 7:55-17:05
REGDAYS = mon-fri
RECORDEXTEN = ""
PARKNOTIFY = SIP/200
OUT_1 = ZAP/g0
OUTPREFIX_1 =
OUTMAXCHANS_1 = 2
OUTCID_1 = 9510512
OPERATOR =
NULL = ""
IN_OVERRIDE = forcereghours
INCOMING = EXT-01
FAX_RX_EMAIL = [email protected]
FAX_RX = system
FAX =
DIRECTORY_OPTS =
DIRECTORY = last
DIAL_OUT = 9
DIAL_OPTIONS = tr
DIALOUTIDS = 1/
CALLFILENAME = ""
AFTER_INCOMING = EXT-01

[ext-did]
include => ext-did-custom
exten => 9510512,1,SetVar(FROM_DID=9510512)
exten => 9510512,2,Goto(ext-local,01,1)

[ext-local]
include => ext-local-custom
exten => 01,1,Macro(exten-vm,novm,01)
exten => 01,hint,SIP/01

[outbound-allroutes]
include => outbound-allroutes-custom
include => outrt-001-ZAP

[outrt-001-ZAP]
include => outrt-001-ZAP-custom
exten => _0X.,1,Macro(dialout-trunk,1,${EXTEN},)
exten => _0X.,2,Macro(outisbusy) ; No available circuits


log:
Mar 17 18:41:02 VERBOSE[3375] logger.c: -- Executing Dial("SIP/01-d015", "ZAP/g0/0800707707") in new stack
Mar 17 18:41:02 NOTICE[3375] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)
Mar 17 18:41:02 VERBOSE[3375] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
Mar 17 18:41:02 DEBUG[3375] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
Mar 17 18:41:02 VERBOSE[3375] logger.c: -- Executing Goto("SIP/01-d015", "s-CONGESTION|1") in new stack
Mar 17 18:41:02 VERBOSE[3375] logger.c: -- Goto (macro-dialout-trunk,s-CONGESTION,1)
Mar 17 18:41:02 VERBOSE[3375] logger.c: -- Executing NoOp("SIP/01-d015", "Dial failed due to CONGESTION") in new stack
Mar 17 18:41:02 VERBOSE[3375] logger.c: -- Executing Macro("SIP/01-d015", "outisbusy") in new stack
Mar 17 18:41:02 VERBOSE[3375] logger.c: -- Executing Playback("SIP/01-d015", "all-circuits-busy-now") in new stack
Mar 17 18:41:02 DEBUG[3375] channel.c: Scheduling timer at 160 sample intervals
Mar 17 18:41:02 VERBOSE[3375] logger.c: -- Playing 'all-circuits-busy-now' (language 'en')
Mar 17 18:41:02 DEBUG[2931] chan_sip.c: Stopping retransmission on '[email protected]' of Response 491505976: Match Found
Mar 17 18:41:04 DEBUG[3375] channel.c: Scheduling timer at 0 sample intervals
Mar 17 18:41:04 DEBUG[3375] channel.c: Scheduling timer at 0 sample intervals
Mar 17 18:41:04 VERBOSE[3375] logger.c: -- Executing Playback("SIP/01-d015", "pls-try-call-later") in new stack
Mar 17 18:41:04 DEBUG[3375] channel.c: Scheduling timer at 160 sample intervals
Mar 17 18:41:04 VERBOSE[3375] logger.c: -- Playing 'pls-try-call-later' (language 'en')


was mache ich falsch?

 

[linuxpingu]

************************************************** ************************************************** *************
Ich versuche es neu mit

A@h 2.7 with Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with "flortz" patch modified by redoctober71
gemäss Anleitung von
http://dondisperato.blogspot.com/
************************************************** ************************************************** *************

;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]


[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=1
callgroup=1
pickupgroup=1
immediate=yes
overlapdial=yes
context=from-pstn
channel => 1-2

;Include genzaptelconf configs
#include zapata-auto.conf

group=2

;Include AMP configs
#include zapata_additional.conf

ergibt log:
Mar 21 10:01:50 VERBOSE[4004] logger.c: -- Executing Dial("SIP/200-b131", "ZAP/g1/0339510512") in new stack
Mar 21 10:01:50 VERBOSE[4004] logger.c: -- Requested transfer capability: 0x00 - SPEECH
Mar 21 10:01:50 VERBOSE[4004] logger.c: -- Called g1/0339510512
Mar 21 10:01:50 DEBUG[3927] chan_zap.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 1
Mar 21 10:01:50 VERBOSE[4004] logger.c: -- Zap/1-1 is proceeding passing it to SIP/200-b131
Mar 21 10:01:50 VERBOSE[3927] logger.c: -- Channel 0/1, span 1 got hangup request
Mar 21 10:01:50 WARNING[4004] app_dial.c: Unable to forward voice
Mar 21 10:01:50 DEBUG[4004] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Mar 21 10:01:50 DEBUG[4004] chan_zap.c: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1
Mar 21 10:01:50 DEBUG[4004] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call
Mar 21 10:01:50 DEBUG[4004] chan_zap.c: disabled echo cancellation on channel 1
Mar 21 10:01:50 DEBUG[4004] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
Mar 21 10:01:50 DEBUG[4004] chan_zap.c: Updated conferencing on 1, with 0 conference users
Mar 21 10:01:50 DEBUG[4004] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Mar 21 10:01:50 DEBUG[4004] chan_zap.c: disabled echo cancellation on channel 1
Mar 21 10:01:50 VERBOSE[4004] logger.c: -- Hungup 'Zap/1-1'
Mar 21 10:01:50 VERBOSE[4004] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
Mar 21 10:01:50 DEBUG[4004] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.
Mar 21 10:01:50 VERBOSE[4004] logger.c: -- Executing Goto("SIP/200-b131", "s-CHANUNAVAIL|1") in new stack
Mar 21 10:01:50 VERBOSE[4004] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
Mar 21 10:01:50 VERBOSE[4004] logger.c: -- Executing NoOp("SIP/200-b131", "Dial failed due to CHANUNAVAIL") in new stack
Mar 21 10:01:50 VERBOSE[4004] logger.c: -- Executing Macro("SIP/200-b131", "outisbusy") in new stack
Mar 21 10:01:50 VERBOSE[4004] logger.c: -- Executing Playback("SIP/200-b131", "all-circuits-busy-now") in new stack
Mar 21 10:01:50 DEBUG[4004] channel.c: Scheduling timer at 160 sample intervals

kann jemand überhaupt über zaphfc ISDN Karte im TE Modus raustelefonieren?
Bei mir gehen leider nur eingehende Rufe.

Besten Dank für jeden Tip

Ivan Tännler