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
-- <SIP/1007-0000084a> 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
username=1007
secret=XXXXX
dtmfmode=rfc2833
host=dynamic
mailbox=1007@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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