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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>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: <o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> -- SIP/voipeer-0000084b redirecting info has changed, passing it to SIP/1007-0000084a<o:p></o:p></p><p class=MsoNormal> -- SIP/voipeer-0000084b is busy<o:p></o:p></p><p class=MsoNormal> == Everyone is busy/congested at this time (1:1/0/0)<o:p></o:p></p><p class=MsoNormal> -- Timeout on SIP/1007-0000084a<o:p></o:p></p><p class=MsoNormal> -- Executing [t@phones:1] Playback("SIP/1007-0000084a", "goodbye") in new stack<o:p></o:p></p><p class=MsoNormal> > 0x7f6a62146640 -- Probation passed - setting RTP source address to 191.96.18.41:62568<o:p></o:p></p><p class=MsoNormal> -- <SIP/1007-0000084a> Playing 'goodbye.slin' (language 'en')<o:p></o:p></p><p class=MsoNormal> > 0x7f6a62146640 -- Probation passed - setting RTP source address to 191.96.18.41:62568<o:p></o:p></p><p class=MsoNormal> -- Executing [t@phones:2] Hangup("SIP/1007-0000084a", "") in new stack<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Sip.conf <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>[1007]<br>type=friend<br>context=sip-phone<br>call-limit=2<br>trustrpid=no<br>callerid="dev1" <1007><br>disallow=all<br>allow=ulaw<br>allow=alaw<br>username=1007<br>secret=XXXXX<br>dtmfmode=rfc2833<br>host=dynamic<br>mailbox=1007@default<br>nat=force_rport,comedia<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Is it a codec issue? Or missed configuration? Asterisk does not know how to translate busy signal. <o:p></o:p></p><p class=MsoNormal>Your help is appreciated!<o:p></o:p></p><p class=MsoNormal>Thanks,<o:p></o:p></p></div></body></html>