Asterisk outgoing call fails with 'Busy Here' error - centos

Question

I have Asterisk 11.7.0 configured on CentOS 6.4 x64 with following sip.conf configuration:
[general]
register =>myuser:pass#voipproviderip
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.0.3
srvlookup=no
subscribecontext=from-sip
[VoIPProvider]
canreinvite=yes
username=myuser
fromuser=myuser
secret=mypass
context=incoming
type=friend
;fromdomain=voipproviderip
host=voipproviderip
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=yes
insecure=very
; ext 100
[100]
type=friend
host=dynamic
secret=MyPass123
context=default
mailbox=100#default
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no
; ext 101
[101]
type=friend
host=dynamic
secret=MyPass123
context=default
mailbox=101#default
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no
; ext 102
[102]
type=friend
host=dynamic
secret=MyPass123
context=default
mailbox=102#default
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no
; ext 103
[103]
type=friend
host=dynamic
secret=MyPass123
context=default
mailbox=103#default
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no
; ext 104
[104]
type=friend
host=dynamic
secret=MyPass123
context=default
mailbox=104#default
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no
; ext 105
[105]
type=friend
host=dynamic
secret=MyPass123
context=default
mailbox=105#default
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no
; ext 106
[106]
type=friend
host=dynamic
secret=MyPass123
context=default
mailbox=106#default
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no
; ext 100
[107]
type=friend
host=dynamic
secret=MyPass123
context=default
mailbox=107#default
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no
; ext 108
[108]
type=friend
host=dynamic
secret=MyPass123
context=default
mailbox=108#default
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no
and following extensions.conf:
[default]
include => internal
include => incoming
include => outgoing
[incoming]
; Ring on extension 100, 200 and the mobile phone.
exten => s,1,Answer()
exten => s,n,Dial(SIP/101&SIP/103,30,r,t,)
; Pass unanswered call to a mobile phone
exten => s,n,Dial(SIP/101&SIP/103/&SIP/100,30,r)
same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
same => n(unavail),VoiceMail(100#default,u)
same => n,Hangup()
same => n(busy),VoiceMail(100#default,b)
same => n,Hangup()
[outgoing]
; Outbound calls can be routed based on the number of digits dialled (or the value of the first few digits)
;exten=> _XXXXXXXXXXXX.,1,Dial(SIP/VoIPProvider/${EXTEN})
;exten=> _XXXXXXXXXXXX.,2,Hangup
exten => _XXXXXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
exten => _XXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
exten => _XXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
exten => _XXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
[internal]
; Calls between employees (between extensions)
exten => _XXX,1,Dial(SIP/${EXTEN},60)
include => outgoing
; Calls to ext 100
exten => 100,1,Dial(SIP/100,20)
exten => 100,n,VoiceMail(100,u)
exten => 100,n,Hangup
; Calls to ext 101
exten => 101,1,Dial(SIP/101,20)
exten => 101,n,VoiceMail(101,u)
exten => 101,n,Hangup
; Calls to ext 102
exten => 102,1,Dial(SIP/102,20)
exten => 102,n,VoiceMail(102,u)
exten => 102,n,Hangup
; Calls to ext 103
exten => 103,1,Dial(SIP/103,20)
exten => 103,n,VoiceMail(103,u)
exten => 103,n,Hangup
; Calls to ext 104
exten => 104,1,Dial(SIP/104,20)
exten => 104,n,VoiceMail(104,u)
exten => 104,n,Hangup
; Calls to ext 105
exten => 105,1,Dial(SIP/105,20)
exten => 105,n,VoiceMail(105,u)
exten => 105,n,Hangup
; Calls to ext 106
exten => 106,1,Dial(SIP/106,20)
exten => 106,n,VoiceMail(106,u)
exten => 106,n,Hangup
; Calls to ext 107
exten => 107,1,Dial(SIP/107,20)
exten => 107,n,VoiceMail(107,u)
exten => 107,n,Hangup
; Calls to ext 108
exten => 108,1,Dial(SIP/108,20)
exten => 108,n,VoiceMail(108,u)
exten => 108,n,Hangup
Whenever I dial a number from CLI>console dial mynumber, I see the folowing output:
-- Executing [mynumber#default:1] Dial("Console/dsp", "SIP/VoIPProvider/mynumber") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/VoIPProvider/mynumber
-- Got SIP response 486 "Busy Here" back from voipproviderip:5060
-- SIP/VoIPProvider-00000000 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Auto fallthrough, channel 'Console/dsp' status is 'BUSY'
I have tried all the possible ways to dial a number, (including / excluding country code, etc.) but it always replays with 486 "Busy Here" could it be that I have to pay the bill of my voip service?

Answer 1

I would try Dial(SIP/mynumber#VoIPProvider);
Also, what do see when you do sip show peers ???