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@phones:1] Playback(“SIP/1007-0000084a”, “goodbye”) in new stack

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

— Playing ‘goodbye.slin’ (language ‘en’)

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

— Executing [t@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 username07
secret=XXXXX
dtmfmode=rfc2833
host=dynamic mailbox07@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,