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:[email protected]/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
The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/2000|30))
andgetaddrinfo("2000|30", "(null)", ...): Name or service not known
. In other words, your config needs to be updated. – Joachim IsakssonDial(SIP/2000|30)
should beDial(SIP/2000,30)
. It's probably generated from the extensions table in the database, the row ` ('from-sip','_X.',1,'Dial','SIP/${EXTEN}|30')` should then be ` ('from-sip','_X.',1,'Dial','SIP/${EXTEN},30')` – Joachim Isaksson