[AsteriskBrasil] Config. E1 ISDN com Siemens Hicom 150

Josué Conti josueconti em gmail.com
Quarta Setembro 13 16:03:50 BRT 2006


Olá Rafael, tudo bem?
Tente isso no zapata.conf

[trunkgroups]

[channels]
context=seu-contexto
switchtype=euroisdn
signalling=pri_cpe
;rxwink=300

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
restrictcid=no

callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes

rxgain=0.0
txgain=0.0
group=1
callgroup=1
immediate=no
callerid=asreceived
musiconhold=sua-musica-em espera
group=1

channel=>1-15
channel=>17-31


Abraço e boa sorte

Josué

2006/9/13, Rafael Augusto <rafael.augusto em govoip.com.br>:
>
> 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
>
>
>
> ----------------------------------------
> Estação VoIP 2006
> 5 e 6 Dezembro
> Curitiba PR
> http://www.estacaovoip.com.br
>
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