I use Asterisk 1.8 with Mysql version 5.1.69 (OS : CentOS Release 6.4, istalled in Virtual machine Oracle VM Virtual Box). From Win 7 32bit, I tried to get information about sip peers using this code :
import java.util.ArrayList;
import java.util.List;
import org.asteriskjava.manager.ManagerConnection;
import org.asteriskjava.manager.ManagerConnectionFactory;
import org.asteriskjava.manager.action.CommandAction;
import org.asteriskjava.manager.response.CommandResponse;
public class Manager
{
private ManagerConnection c;
public Manager() throws Exception
{
ManagerConnectionFactory factory = new ManagerConnectionFactory(
"172.16.213.91", "asterisk", "asterisk");
c = factory.createManagerConnection();
}
public void run() throws Exception
{
c.login();
CommandAction action;
CommandResponse response;
List<String> list = new ArrayList<String>();
action = new CommandAction();
//action.setCommand("core show help");
action.setCommand(" sip show peers");
response = (CommandResponse) c.sendAction(action);
list = response.getResult();
System.out.println(list.size());
int i = 0;
while ( i <list.size())
{
System.out.println(list.get(i));
i++;
}
c.logoff();
}
public static void main(String[] args) throws Exception
{
new Manager().run();
}
}I got this output in console :
18 avr. 2014 11:01:07 org.asteriskjava.manager.internal.ManagerConnectionImpl connect
INFO: Connecting to 172.16.213.91:5038
18 avr. 2014 11:01:07 org.asteriskjava.manager.internal.ManagerConnectionImpl setProtocolIdentifier
INFO: Connected via Asterisk Call Manager/1.1
18 avr. 2014 11:01:07 org.asteriskjava.manager.internal.ManagerConnectionImpl setProtocolIdentifier
ATTENTION: Unsupported protocol version 'Asterisk Call Manager/1.1'. Use at your own risk.
18 avr. 2014 11:01:07 org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
INFO: Successfully logged in
18 avr. 2014 11:01:07 org.asteriskjava.manager.internal.EventBuilderImpl buildEvent
INFO: No event class registered for event type 'fullybooted', attributes: {status=Fully Booted, event=FullyBooted, privilege=system,all}
18 avr. 2014 11:01:09 org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
INFO: Determined Asterisk version: Asterisk 1.0
18 avr. 2014 11:01:09 org.asteriskjava.manager.internal.ManagerConnectionImpl disconnect
INFO: Closing socket.
2
Name/username Host Dyn Forcerport ACL Port Status Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]Even, I connected to Asterisk with sip user. I got the same result.
I have 2 sip users in my Data base . I use asterisk-java-0.3.jar in my project.
I tried also with telnet this command from CentOS:
action: sippeers
actionid: 4I got this response:
Response: Success
ActionID: 4
EventList: start
Message: Peer status list will follow
Event: PeerlistComplete
EventList: Complete
ListItems: 0
ActionID: 4From Cli, if there is not any peer connected, I got this massage
[root@Acer5635 ~]# asterisk -cvvvv
Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect.
[root@Acer5635 ~]# asterisk -rcvvvv
Asterisk 1.8.22.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.8.22.0 currently running on Acer5635 (pid = 3750)
Verbosity is at least 4
Acer5635*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
Acer5635*CLI>My manager.conf is like this:
[general]
enabled = yes
webenabled = yes
port = 5038
bindaddr = 0.0.0.0
[asterisk]
secret = asterisk
permit = 0.0.0.0/0.0.0.0
read = all,system,call,log,verbose,command,agent,user,config
write = all,system,call,log,verbose,command,agent,user,configextensions.conf:
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=DAHDI/G2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[dundi-e164-customers]
[dundi-e164-via-pstn]
[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
switch => DUNDi/e164
[dundi-e164-lookup]
include => dundi-e164-local
include => dundi-e164-switch
[macro-dundi-e164]
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
[iaxprovider]
[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
[trunkld]
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunklocal]
exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunktollfree]
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[international]
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
ignorepat => 9
include => local
include => trunkld
[local]
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
[outbound-freenum]
exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
[outbound-freenum2]
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ; make sure the suffix is all digits as well
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
same => n,Set(TIMEOUT(absolute)=10800)
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; perform our lookup with freenum.org
same => n,GotoIf($["${isnresult}" != ""]?from)
same => n,Set(DIALSTATUS=CONGESTION)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we'll use it for our outbound dialing domain
same => n(dial),Dial(SIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)
exten => fn-BUSY,1,Busy()
exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()
[macro-trunkdial]
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp
[stdexten]
exten => _X.,50000(stdexten),NoOp(Start stdexten)
exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
exten => _X.,n,Set(LOCAL(dev)=${ARG1})
exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
exten => _X.,n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return()
[stdPrivacyexten]
exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
exten => _X.,n,Set(LOCAL(ext)=${ARG1})
exten => _X.,n,Set(LOCAL(dev)=${ARG2})
exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
exten => _X.,n,Set(LOCAL(cntx)=${ARG5})
exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call _X.creening)
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1) ; Callee chose to send this call to a polite "Don't call again" script.
exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1) ; Callee chose to send this call to a telemarketer torture script.
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return
[macro-page];
exten => s,1,ChanIsAvail(${ARG1},s) ; s is for ANY call
exten => s,n,GoToIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1})
exten => s,n(fail),Hangup
[demo]
include => stdexten
exten => s,1,Wait(1) ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." (but skip if channel is not up)
exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
exten => 1234,n,Goto(default,s,1) ; exited Voicemail
exten => 1235,1,Voicemail(1234,u) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(1234,b) ; Unless busy
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
exten => 76245,1,Macro(page,SIP/Grandstream1)
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})
[default]
include => demo
[time]
exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
exten => _X.,n,Wait(0.25)
exten => _X.,n,Answer()
exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
exten => _X.,n,SayPhonetic(z)
exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
exten => _X.,n,Playback(spy-local)
exten => _X.,n,WaitUntil(${FUTURETIME})
exten => _X.,n,Playback(beep)
exten => _X.,n,Return()
[ani]
exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
exten => _X.,n,Wait(0.25)
exten => _X.,n,Answer()
exten => _X.,n,Playback(vm-from)
exten => _X.,n,SayDigits(${CALLERID(ani)})
exten => _X.,n,Wait(1.25)
exten => _X.,n,SayDigits(${CALLERID(ani)}) ; playback again in case of missed digit
exten => _X.,n,Return()
[internal]
switch => Realtime/internal@realtime_extextconfig.conf :
[settings]
sipusers => mysql,asterisk,users1 ; SIP user
sippeers => mysql,asterisk,users1 ; SIP peers
extensions => mysql,asterisk,extensions ; SIP extensions
voicemail => mysql,asterisk,voicemails ; SIP voicemailboxes
queues => mysql,asterisk,queues ; SIP queue
queue_members => mysql,asterisk,queue_members ; SIP queue members res_config_mysql.conf :
[asterisk]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = asterisk
dbpass = asterisk
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
requirements=warn ; or createclose or createcharmodules.conf :
[modules]
autoload=yes
noload => pbx_gtkconsole.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_console.soI don’t know from where comes the problem. I need the java class to build a Swing java program that get me information about sip peers