title: Asterisk 11 cannot make a call after installing MySQL real-time database
body: <p>I've installed Asterisk 11 on CentOS 6.4 and MySQL Database real-time. <a href="http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example" rel="nofollow">I followed this manual</a>. </p>
  
  <p>I changed all sip.conf, extenstions.conf and extconfig.conf. I could successfully connect Asterisk to MySQL database in real-time. User registration worked fine. However, I couldn't hear any sound from the phone, and I couldn't make a call to another phone.</p>
  
  <p>here is the logs when I make a call to "user 2000" from "user 1000"</p>
  
  <pre><code>[Apr 12 00:24:11] WARNING[19656][C-0000000c]: sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead
  [Apr 12 00:24:11] WARNING[19656][C-0000000c]: sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead
  [Apr 12 00:24:11] WARNING[19656][C-0000000c]: sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead
    == Using SIP RTP CoS mark 5
  [Apr 12 00:24:11] WARNING[19656][C-0000000c]: sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead
      -- Executing [2000@from-sip:1] Dial("SIP/1000-0000000b", "SIP/2000|30")
  [Apr 12 00:24:11] WARNING[22592][C-0000000c]: pbx.c:1618 pbx_exec: The application delimiter is now the comma, not the pipe.  Did you forget to convert your dialplan?  (Dial(SIP/2000|30))
    == Using SIP RTP CoS mark 5
  [Apr 12 00:24:11] ERROR[22592][C-0000000c]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("2000|30", "(null)", ...): Name or service not known
  [Apr 12 00:24:11] WARNING[22592][C-0000000c]: chan_sip.c:6204 create_addr: No such host: 2000|30
  [Apr 12 00:24:11] WARNING[22592][C-0000000c]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
    == Everyone is busy/congested at this time (1:0/0/1)
      -- Executing [2000@from-sip:2] VoiceMail("SIP/1000-0000000b", "2000@from-sip")
      -- &lt;SIP/1000-0000000b&gt; Playing 'vm-intro.gsm' (language 'en')
      -- &lt;SIP/1000-0000000b&gt; Playing 'beep.gsm' (language 'en')
      -- Recording the message
      -- x=0, open writing:  /var/spool/asterisk/voicemail/from-sip/2000/tmp/5lOmnn format: wav49, 0x7fa2b80013d8
      -- x=1, open writing:  /var/spool/asterisk/voicemail/from-sip/2000/tmp/5lOmnn format: gsm, 0x7fa2b80049b8
      -- x=2, open writing:  /var/spool/asterisk/voicemail/from-sip/2000/tmp/5lOmnn format: wav, 0x7fa2b80041b8
  </code></pre>
  
  <p>Do you see what is the problem?</p>
  
  <p><strong>sip.conf</strong></p>
  
  <pre><code>[general]
   context=from-sip                        ; Default context for incoming calls
   ;
   bindport=5060                   ; bindport is the local UDP port that Asterisk will listen on
   bindaddr=0.0.0.0           ; IP address to bind to (0.0.0.0 binds to all)
   ;
   disallow=all                    ; First disallow all codecs
   allow=gsm
   allow=ulaw                      ; Allow codecs in order of preference
   ;
   register =&gt; 12121111111:1234:11111111@sipauth.deltathree.com/1000
  </code></pre>
  
  <p><strong>extensions.conf</strong></p>
  
  <pre><code>[general]
   [globals]
   ;
   [from-sip]
   switch =&gt;Realtime
  </code></pre>
  
  <p><strong>res_config_mysql.conf</strong></p>
  
  <pre><code>[general]
  dbhost = 127.0.0.1
  dbname = asteriskrealtime
  dbuser = root
  dbpass = password
  dbport = 3306
  </code></pre>