From: Arthur R. <art...@gm...> - 2015-03-18 22:02:45
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Hello Again, Nailed one of those : Second, and a very important one too. *I do not understand how asterisk defines if it is "from-openbts" or "from-pstn" or "phones" ?* *Simply into sip.conf and sip-*.conf* *I need to look if my from pstn is a zaptel.* *Still, I cannot get issue 1 from my previous mail. Other issues are a bonus.* *Thanks* Le mer. 18 mars 2015 à 21:28, Arthur Rabaté <art...@gm...> a écrit : > Hello, > > I ran into a couple issues with asterisk & openbts. > > First and most important,* i cannot get my phones to ring for more than > 30 seconds*. > I can set in the extension file a 10 secs ringing (yet this starts when > the call is initiated, not when the phone starts ringing - hope this is > clear) > > This is a sample log from asterisk. > Extract from asterisk extensions.conf : > [to-openBTS](HangupCause) ;GoSub for OpenBTS users > ..... > same => n,Set(CALLERID(name)=${CDR(A-Number)}) > same => n,Dial(SIP/${ARG1}@${ARG2}:${ARG3},60) > > As you can see I have configured the ringing time to 60s > > And the log : > [...] > > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [5002@to-openBTS:6] Set("SIP/20896100010-00000006", > "CDR(hangupdirection)=SYSTEM") in new stack > -- Executing [5002@to-openBTS:7] GotoIf("SIP/20896100010-00000006", > "0?Voicemail,5002,1") in new stack > -- Executing [5002@to-openBTS:8] GotoIf("SIP/20896100010-00000006", > "0?:h-18,1") in new stack > -- Goto (to-openBTS,h-18,1) > -- Executing [h-18@to-openBTS:1] Progress("SIP/20896100010-00000006", > "") in new stack > -- Executing [h-18@to-openBTS:2] NoOp("SIP/20896100010-00000006", " > *AST_CAUSE_NO_USER_RESPONSE,noanswer*") in new stack > -- Executing [h-18@to-openBTS:3] Hangup("SIP/20896100010-00000006", > "18") in new stack > == Spawn extension (to-openBTS, h-18, 3) exited non-zero on > 'SIP/20896100010-00000006' > -- Executing [h@to-openBTS:1] Log("SIP/20896100010-00000006", > "NOTICE,A-Number=5001 A-Name= A-IMSI=IMSI208960000005001 B-Number=5002 > B-Name= B-IMSI=IMSI208960000005002 hangupcause=18 *dialstatus=CHANUNAVAIL > hangupdirection=SYSTEM duration=33* billsec=0") > > Duration is 33s and I get a *chanunavail* ... > > If I set up 10 secs: > same => n,Dial(SIP/${ARG1}@${ARG2}:${ARG3},10) > > I get a 10second duration and noanswer : > > > *[...]*("*SIP/20896100010-00000012*", "*AST_CAUSE_UNSPECIFIED,noanswer*" > [...] > *dialstatus=NOANSWER hangupdirection=SYSTEM duration=10* > > > Second, and a very important one too. *I do not understand how asterisk > defines if it is "from-openbts" or "from-pstn" or "phones" ?* > > I would like to eventually add a special context for sip phones (I know I > could simply add them to the default) > > > A less important one is vm. VoiceMails are not configured by default, you > need to add them manually. So I added these simple lines : > [default](+) ;Here you can add any dialplan the phones must be able to > call internaly > exten => 111, > 1,ExecIf(${VM_INFO(${CDR(A-Number)},exists)}?Goto(VoicemailMain,${CDR(A-Number)},1):Goto(generatevm,s,1)},1) > ; if not configured we go to generatevm > > [generatevm] > exten => s, 1,Set(CDR(hangupdirection)=A) > same => n,Exec(${SHELL(/OpenBTS/OpenBTSCLI -c sendsms > ${FILTER(0-9,${CDR(A-IMSI)})} 111 'Messagerie configuree : rappelez 111' >> > /dev/null)}) > same => n,Exec(${SHELL(echo "\n${CDR(A-Number)} =>" >> > /etc/asterisk/voicemail-customer.conf)},${SHELL(reload asterisk)}) > same => n,Set(CDR(hangupdirection)=SYSTEM) > same => n,Hangup(16) > > It sets up the vm when the first call to the vm is initiated, and reloads > the asterisk conf. > *This is far from perfect so I would rather have an "configured for > everyone" voicemail. Is there a way to do this ?* > > > Last : I have not taken the time to look into this. When a softphone calls > a gsm, the caller id is 0000. And the gsm cannot call the softphone because > asterisk redirects it to the to-pstn exten. > > > If anyone can give some insight , well, thanks ! > > Regards, > > Arthur > > > |