[asterisk-bugs] [JIRA] (ASTERISK-28867) cannot get ANSWER Status from ${DIALSTATUS} though i get busy congested

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28867?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team closed ASTERISK-28867.
------------------------------------

    Resolution: Not A Bug

> cannot get ANSWER Status from ${DIALSTATUS} though i get busy congested 
> ------------------------------------------------------------------------
>
>                 Key: ASTERISK-28867
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28867
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: Applications/app_amd, CDR/cdr_manager
>    Affects Versions: 13.28.1
>         Environment: vicidial
>            Reporter: irfan shafi
>
> exten => _XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
> exten => _XXXXXXXX,n,Set(UniqueID=${UNIQUEID})
> exten => _XXXXXXXX,n,Set(CALLED=${EXTEN})
> exten => _XXXXXXXX,n,Set(DID=${CALLERID(num)})
> exten => _XXXXXXXX,n,NoOp(============ ${UniqueID} ==============)
> exten => _XXXXXXXX,n,Set(CALLTIME=${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)})
> exten => _XXXXXXXX,n,Set(CALLFILE=${CALLTIME}_${DID}_${CALLED})
> exten => _XXXXXXXX,n,Set(CALLTYPE=outbound)
> exten => _XXXXXXXX,n,MixMonitor(/var/spool/asterisk/monitorDONE/MP3/${CALLFILE}.wav)
> ;exten => _XXXXXXXX,n,MixMonitor(/var/spool/asterisk/clicktocall/${CALLFILE}.wav)
> exten => _XXXXXXXX,n,AGI(recording_log.php,${CALLFILE},${DID},${UniqueID},${Caller},${CALLED})
> exten => _XXXXXXXX,n,Dial(SIP/${EXTEN}@zain,40,tTo)
> exten => _XXXXXXXX,n,NoOp(============ dial status is ${DIALSTATUS} by irfan ==============)
> exten => _XXXXXXXX,n,Goto(s-${DIALSTATUS},1)
> exten => s-ANSWER,1,Goto(connected,1)
> exten => s-ANSWERED,1,Goto(connected,1)
> exten => s-CONGESTION,1,Goto(notconnected,1)
> exten => s-NOANSWER,1,Goto(misscall,1)
> exten => s-BUSY,1,Goto(notconnected,1)
> exten =>misscall,1,AGI(misscall_log.php,${CALLFILE},${DID},${UniqueID},${Caller},${CALLED})
> exten =>misscall,2,AGI(CallLogApi.php,${DID},${Caller},${CALLED},2)
> exten => _XXXXXXXX,n,NoOp(============ This was a misscall ==============);
> exten =>connected,1,AGI(CallLogApi.php,${DID},${Caller},${CALLED},1)
> exten => _XXXXXXXX,n,NoOp(============ This was a misscall ==============);
> exten =>notconnected,1,AGI(CallLogApi.php,${DID},${Caller},${CALLED},0)
> exten => _XXXXXXXX,n,NoOp(============ This was a misscall ==============);
> ;exten => _XXXXXXXX,n,Hangup
> i can print the status of call if its busy , no answer , congested 
> but cannot do it for answered calls even though it displays on asterisk cli thats call is answered 
> Asterisk CLI output is below 
> [May  4 15:53:11]   == Begin MixMonitor Recording SIP/zain-0000001c
> [May  4 15:53:11]     -- Launched AGI Script /usr/share/asterisk/agi-bin/recording_log.php
> [May  4 15:53:11]     -- <SIP/zain-0000001c>AGI Script recording_log.php completed, returning 0
> [May  4 15:53:11]     -- Executing [67771404 at 1clicktocall:11] Dial("SIP/zain-0000001c", "SIP/67771404 at zain,40,tTo") in new stack
> [May  4 15:53:11]   == Using SIP RTP CoS mark 5
> [May  4 15:53:11]     -- Called SIP/67771404 at zain
> [May  4 15:53:11]        > 0x7f249401c7c0 -- Probation passed - setting RTP source address to 192.168.86.7:21810
> [May  4 15:53:11]        > 0x7f249401c7c0 -- Probation passed - setting RTP source address to 192.168.86.7:21810
> [May  4 15:53:11]     -- SIP/zain-0000001d is making progress passing it to SIP/zain-0000001c
> [May  4 15:53:11]        > 0x7f249401c7c0 -- Probation passed - setting RTP source address to 192.168.86.7:21810
> [May  4 15:53:11]     -- SIP/zain-0000001d is making progress passing it to SIP/zain-0000001c
> [May  4 15:53:11]        > 0x7f249401c7c0 -- Probation passed - setting RTP source address to 192.168.86.7:21810
> [May  4 15:53:12]     -- SIP/zain-0000001d is making progress passing it to SIP/zain-0000001c
> [May  4 15:53:12]        > 0x7f249401c7c0 -- Probation passed - setting RTP source address to 192.168.86.7:21810
> [May  4 15:53:18]     -- SIP/zain-0000001d answered SIP/zain-0000001c



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