简体   繁体   中英

Asterisk 11 cannot make a call after installing MySQL real-time database

I've installed Asterisk 11 on CentOS 6.4 and MySQL Database real-time. I followed this manual .

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.

here is the logs when I make a call to "user 2000" from "user 1000"

[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")
    -- <SIP/1000-0000000b> Playing 'vm-intro.gsm' (language 'en')
    -- <SIP/1000-0000000b> 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

Do you see what is the problem?

sip.conf

[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 => 12121111111:1234:11111111@sipauth.deltathree.com/1000

extensions.conf

[general]
 [globals]
 ;
 [from-sip]
 switch =>Realtime

res_config_mysql.conf

[general]
dbhost = 127.0.0.1
dbname = asteriskrealtime
dbuser = root
dbpass = password
dbport = 3306

You have write correct dialplan in extensions table in db.

For asterisk below 1.4 you have use appdata with pipe (|) delimiter, for newer version you have use comma.

The technical post webpages of this site follow the CC BY-SA 4.0 protocol. If you need to reprint, please indicate the site URL or the original address.Any question please contact:yoyou2525@163.com.

 
粤ICP备18138465号  © 2020-2024 STACKOOM.COM