[AsteriskBrasil] Config. E1 ISDN com Siemens Hicom 150

[Rafael Augusto]

 Dio, segue o erro ao efetuar o dial.


Executing Macro("SIP/200-0a0fbac0", "dialout-trunk|2|100|") in new stack
    -- Executing GotoIf("SIP/200-0a0fbac0", "1?3:2)") in new stack
    -- Goto (macro-dialout-trunk,s,3)
    -- Executing Macro("SIP/200-0a0fbac0", "user-callerid") in new stack
    -- Executing DBget("SIP/200-0a0fbac0", "AMPUSER=DEVICE/200/user") in new
stack
    -- DBget: varname=AMPUSER, family=DEVICE, key=200/user
    -- DBget: set variable AMPUSER to 200
    -- Executing DBget("SIP/200-0a0fbac0",
"AMPUSERCIDNAME=AMPUSER/200/cidname") in new stack
    -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=200/cidname
    -- DBget: set variable AMPUSERCIDNAME to 200
    -- Executing GotoIf("SIP/200-0a0fbac0", "0?5") in new stack
    -- Executing SetCallerID("SIP/200-0a0fbac0", ""200" <200>") in new stack
    -- Executing NoOp("SIP/200-0a0fbac0", "Using CallerID "200" <200>") in
new stack
    -- Executing Macro("SIP/200-0a0fbac0", "record-enable|200|OUT") in new
stack
    -- Executing GotoIf("SIP/200-0a0fbac0", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing AGI("SIP/200-0a0fbac0",
"recordingcheck|20060913-143611|1158172571.7") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20060913-143611|1158172571.7: Outbound recording not
enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/200-0a0fbac0", "No recording needed") in new
stack
    -- Executing Macro("SIP/200-0a0fbac0", "outbound-callerid|2") in new
stack
    -- Executing DBget("SIP/200-0a0fbac0",
"USEROUTCID=AMPUSER/200/outboundcid") in new stack
    -- DBget: varname=USEROUTCID, family=AMPUSER, key=200/outboundcid
    -- DBget: set variable USEROUTCID to 200
    -- Executing GotoIf("SIP/200-0a0fbac0", "0?4") in new stack
    -- Executing SetCallerID("SIP/200-0a0fbac0", "Rota PABX") in new stack
    -- Executing GotoIf("SIP/200-0a0fbac0", "0?6") in new stack
    -- Executing SetCallerID("SIP/200-0a0fbac0", "200") in new stack
    -- Executing NoOp("SIP/200-0a0fbac0", "CallerID set to 200") in new
stack
    -- Executing SetGroup("SIP/200-0a0fbac0", "OUT_2") in new stack
    -- Executing CheckGroup("SIP/200-0a0fbac0", "30") in new stack
    -- Executing SetVar("SIP/200-0a0fbac0", "DIAL_NUMBER=100") in new stack
    -- Executing SetVar("SIP/200-0a0fbac0", "DIAL_TRUNK=2") in new stack
    -- Executing AGI("SIP/200-0a0fbac0", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing SetVar("SIP/200-0a0fbac0", "OUTNUM=100") in new stack
    -- Executing Cut("SIP/200-0a0fbac0", "custom=OUT_2|:|1") in new stack
    -- Executing GotoIf("SIP/200-0a0fbac0", "0?16") in new stack
    -- Executing Dial("SIP/200-0a0fbac0", "ZAP/g1/100") in new stack
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Goto("SIP/200-0a0fbac0", "s-CONGESTION|1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing NoOp("SIP/200-0a0fbac0", "Dial failed due to CONGESTION")
in new stack
    -- Executing Macro("SIP/200-0a0fbac0", "outisbusy") in new stack
    -- Executing PlayTones("SIP/200-0a0fbac0", "Busy") in new stack
    -- Executing Macro("SIP/200-0a0fbac0", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/200-0a0fbac0", "w") in new stack
    -- Executing NoCDR("SIP/200-0a0fbac0", "") in new stack
    -- Executing Wait("SIP/200-0a0fbac0", "5") in new stack
    -- Executing Hangup("SIP/200-0a0fbac0", "") in new stack

Abraços,

Rafael







Message: 2
Date: Wed, 13 Sep 2006 11:35:07 -0300 (ART)
From: Dio Makibara <dioedu em yahoo.com.br>
Subject: Re: [AsteriskBrasil] Config. E1 ISDN com Siemens Hicom 150
To: Rafael Augusto <rafael_jcn em yahoo.com.br>
Cc: Asterisk Brasil <asteriskbrasil em listas.asteriskbrasil.org>
Message-ID: <20060913143507.3048.qmail em web52504.mail.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"

Rafael,
 
 Envie as mensagens para lista, pois a chance de ser respondida é maior.
 
 Mas aparentemente as configurações estão corretas. Informe o que está sendo
exibido no console do asterisk ao tentar efetuar uma ligação.
 
 Diógenes Makibara
 

Rafael Augusto <rafael_jcn em yahoo.com.br> escreveu: Dio, segue configuração
do zaptel e zapata, o extensions é tranquilo, se poder me ajudar, desde de
já agradeço.
   
  zaptel.conf
  span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31

  loadzone        = br
defaultzone     = br

  zapata.conf
   
  [channels]
  language=br
context=from-pstn
signalling=fxs_ks
rxwink=300              ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO  lines
;
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
group=1
   
  group = 1
context =ext-ddr
signalling=pri_net
overlapdial=yes
immediate=no
channel => 1-15
channel => 17-31

   
  Connected to Asterisk 1.2.10 currently running on govoip (pid = 19866)
Verbosity is at least 3
govoip*CLI> pri  show span 1
Primary D-channel: 16
Status: Provisioned, In Alarm, Down, Active
Switchtype: National ISDN
Type: Network
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: -1
T200 Timer: 1000
T203 Timer: 10000
T305 Timer: 30000
T308 Timer: 4000
T313  Timer: 4000
N200 Counter: 3

  Agraços,
   
  Rafael