asterisk users - Feb 2017 - asterisk 13.13.1 Everyone is busy-congested at this time (1:1/0/0)

Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My
extensions.conf file was mostly copied from server running Asterisk 1.8.
That being said! If I dial a number and get a busy signal I get the
following error: 

 

    -- SIP/voipeer-0000084b redirecting info has changed, passing it to
SIP/1007-0000084a

    -- SIP/voipeer-0000084b is busy

  == Everyone is busy/congested at this time (1:1/0/0)

    -- Timeout on SIP/1007-0000084a

    -- Executing [t at phones:1] Playback("SIP/1007-0000084a",
"goodbye") in
new stack

       > 0x7f6a62146640 -- Probation passed - setting RTP source address to
191.96.18.41:62568

    -- <SIP/1007-0000084a> Playing 'goodbye.slin' (language
'en')

       > 0x7f6a62146640 -- Probation passed - setting RTP source address to
191.96.18.41:62568

    -- Executing [t at phones:2] Hangup("SIP/1007-0000084a",
"") in new stack

 

Sip.conf 

[1007]
type=friend
context=sip-phone
call-limit=2
trustrpid=no
callerid="dev1" <1007>
disallow=all
allow=ulaw
allow=alaw
username=1007
secret=XXXXX
dtmfmode=rfc2833
host=dynamic
mailbox=1007 at default
nat=force_rport,comedia

 

Is it a codec issue? Or missed configuration? Asterisk does not know how to
translate busy signal. 

Your help is appreciated!

Thanks,

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