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Summary:ASTERISK-28867: cannot get ANSWER Status from ${DIALSTATUS} though i get busy congested
Reporter:irfan shafi (irfanshafi20)Labels:
Date Opened:2020-05-04 08:21:12Date Closed:2020-05-04 08:21:13
Priority:MajorRegression?
Status:Closed/CompleteComponents:Applications/app_amd CDR/cdr_manager
Versions:13.28.1 Frequency of
Occurrence
Related
Issues:
Environment:vicidialAttachments:
Description:exten => _XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXX,n,Set(UniqueID=${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@1clicktocall:11] Dial("SIP/zain-0000001c", "SIP/67771404@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@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
Comments:By: Asterisk Team (asteriskteam) 2020-05-04 08:21:13.344-0500

We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

If this issue is actually a bug please use the Bug issue type instead.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
By: Asterisk Team (asteriskteam) 2020-05-04 08:21:14.682-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.