[AsteriskBrasil] Problemas nas ligações PSTN e via Provedor

[Fabricio Louback]

Prezada lista,

Estou fazendo um laboratório básico e estou com o seguinte cenário:

- Um servidor com SUSE10, Asterisk 1.2.1, MD3200 e um X-Lite
- Um PC rodando X-Lite

O asterisk já esta rodando e os X-lites se falam internamente.

Problema 01: Tenho uma conta na voiceconnect, sendo que quando digite
"sip show peers" mostra a conta na VC autenticada perfeitamente, mas
não estou conseguindo fazer ligações   apartir do X-lIte utilizando a
VOICE CONNECT.

Problema 02: Configurei o MD3200 e também esta ok, só que não consegui
efetuar ligaões atraves dos X-Lites para a PSTN utilizando a MD3200.
Estou usando a MD3200 conectada direto em um linha PSTN.

Segue abaixo meu EXTENSIONS.CONF e SIP.CONF, se alguém puder me ajudar.

##################SIP.CONF#######################

[default]
context=chamadaentrante
disallow=all
allow=gsm
allow=ulaw
allow-alaw
allow=ilbc
insecure=very
register => 29930xx:xxxxxx em sip.voiceconnect.com.br:5060

[200]
username=200
type=friend
secret=200
qualify=yes
port=5060
nat=never
host=dynamic
dtmfmode=rfc2833
context=ramais
canreinvite=no
callerid="200"
disallow=all
allow=gsm
allow=ulaw
allow-alaw
allow=ilbc

[201]
username=201
type=friend
secret=201
qualify=yes
port=5060
nat=never
host=dynamic
dtmfmode=rfc2833
context=ramais
canreinvite=no
callerid="201"
disallow=all
allow=gsm
allow=ulaw
allow-alaw
allow=ilbc


[voiceconnect]
type=peer
username=29930xx
fromuser=29930xx
secret=xxxxxxxxxx ; password used to login their website
host=sip.voiceconnect.com.br
nat=yes ; my asterisk is behind nat
canreinvite=yes
qualify=yes
context=inbound-voiceconnect ; context to be used in extensions.conf
for inbound calls from voiceconnect


##############EXTENSIONS.CONF############################

[chamadaentrante]
exten=>s,1,Dial(SIP/201)
exten=>s,2,Congestion
exten=>s,3,Hangup
exten=>s,102,Congestion
exten=>s,103,Hangup

h,1,hangup

[ramais]

exten=>200,1,dial(SIP/200)
exten=>200,2,Congestion
exten=>200,3,Hangup
exten=>200,102,Congestion
exten=>200,103,Hangup

exten=>201,1,dial(SIP/201)
exten=>201,2,Congestion
exten=>201,3,Hangup
exten=>201,102,Congestion
exten=>201,103,Hangup


[inbound-voiceconnect]
 exten=>201,1,Wait(1) ;Wait a second, and to get the CallerID information.
 exten=>201,2,Playback(support) ; play initial greeting/recording
 exten=>201,3,Dial(${OFFICE},20,tr) ; dial my sip phone for 20 sec
 exten=>201,4,SetCallerID(${IDCALLER}) ; im not there? set call id to
pass to my cell phone
 exten=>201,5,Dial(IAX2/login em voipjet/cellphone#,10,tr) ; dial my cell
phone , and seamlessly fw call to it. I use voipjet Smile
 exten=>201,6,Playback(busy-on-call) ; switch to a on-call message if
i didnt pick up both cell and sip phone
 exten=>201,7,Voicemail(1) ;leave voice mail
 exten=>201,8,Hangup() ; hangup the call

[saida-pstn]

exten=>_.,1,Dial(Zap/g1/${EXTEN:1},45,Tt)
exten=>_.,2,Hangup


h,1,hangup

[voiceconnect]
exten=>_010,1,SetCIDNNum(29930xx)
exten=>_010,2,SetCIDNName(29930xx)
exten=>_010,3,Dial(SIP/${EXTEN:3}@voiceconnect)
exten=>_010,4,Playback(invalid)
exten=>_010,5,Hangup