[asterisk-bugs] [JIRA] (ASTERISK-25328) Wrong ANSWEREDTIME after Dial app execution

[Richard Mudgett (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25328?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227306#comment-227306 ] 

Richard Mudgett commented on ASTERISK-25328:
--------------------------------------------

The ANSWEREDTIME starts when the channel is answered.  Don't manually answer the call before dialing.  When the called channel answers it will automatically answer the calling channel so they can be bridged.  This way the ANSWEREDTIME will also be the bridged time.  You should not need to manually answer the call unless your dialplan is going to interact with the caller such as an automated attendent.

> Wrong ANSWEREDTIME after Dial app execution
> -------------------------------------------
>
>                 Key: ASTERISK-25328
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25328
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_dial
>    Affects Versions: 13.1.0
>            Reporter: Sébastien M.
>            Assignee: Unassigned
>         Attachments: debugANSWEREDTIME-Asterisk11, debugANSWEREDTIME-Asterisk13
>
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial
> ??DIALEDTIME - This is the time from dialing a channel until when it is disconnected.??
> ??ANSWEREDTIME - This is the amount of time for actual call.??
> ??DIALSTATUS - This is the status of the call??
> *Dialplan :*
> {quote}
> exten => _017013xxxx,1,Answer()
> exten => _017013xxxx,n,wait(5)
> exten => _017013xxxx,n,Dial(SIP/027273xxxx at CBV2-BCT1-VIP-MGC,20,gH)
> exten => _017013xxxx,n,NoOp(DIALSTATUS is $\{DIALSTATUS\}, ANSWEREDTIME is $\{ANSWEREDTIME\}, DIALEDTIME is $\{DIALEDTIME\})
> {quote}
> *Test :*
> I call 017013xxxx. After 5 second, Asterisk dial the 027273xxxx.
> The 027273xxxx waits for 4 seconds and pick up the call for 10 seconds before hang up.
> $\{ANSWEREDTIME\} should be 10.
> *Log :*
> {quote}
>  == Using SIP RTP CoS mark 5
>     -- Executing \[017013xxxx at public:1\] Answer("SIP/CBV2-BCT1-VIP-MGC-0000002b", "") in new stack
>     -- Executing \[017013xxxx at public:2\] Wait("SIP/CBV2-BCT1-VIP-MGC-0000002b", "5") in new stack
>     -- Executing \[017013xxxx at public:3\] Dial("SIP/CBV2-BCT1-VIP-MGC-0000002b", "SIP/027273xxxx at CBV2-BCT1-VIP-MGC,20,gH") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/027273xxxx at CBV2-BCT1-VIP-MGC
>     -- SIP/CBV2-BCT1-VIP-MGC-0000002c is ringing
>     -- SIP/CBV2-BCT1-VIP-MGC-0000002c is making progress passing it to SIP/CBV2-BCT1-VIP-MGC-0000002b
>        > 0x2b554d6328d0 -- Probation passed - setting RTP source address to 46.255.54.7:32102
>        > 0x2b5570029720 -- Probation passed - setting RTP source address to 46.255.54.5:35774
>     -- SIP/CBV2-BCT1-VIP-MGC-0000002c answered SIP/CBV2-BCT1-VIP-MGC-0000002b
>     -- Channel SIP/CBV2-BCT1-VIP-MGC-0000002b joined 'simple_bridge' basic-bridge <f79aafa2-149a-49d0-b13b-1c3dee8c5db0>
>     -- Channel SIP/CBV2-BCT1-VIP-MGC-0000002c joined 'simple_bridge' basic-bridge <f79aafa2-149a-49d0-b13b-1c3dee8c5db0>
>     -- Channel SIP/CBV2-BCT1-VIP-MGC-0000002c left 'simple_bridge' basic-bridge <f79aafa2-149a-49d0-b13b-1c3dee8c5db0>
>     -- Channel SIP/CBV2-BCT1-VIP-MGC-0000002b left 'simple_bridge' basic-bridge <f79aafa2-149a-49d0-b13b-1c3dee8c5db0>
>     -- Executing \[017013xxxx at public:4\] NoOp("SIP/CBV2-BCT1-VIP-MGC-0000002b", "DIALSTATUS is ANSWER, ANSWEREDTIME is 19, DIALEDTIME is 19") in new stack
>     -- Auto fallthrough, channel 'SIP/CBV2-BCT1-VIP-MGC-0000002b' status is 'ANSWER'
> {quote}
> *Result :*
> $\{ANSWEREDTIME\} is 19 but should be 10.
> On Asterisk 11.9.0, ANSWEREDTIME works well and was 10.



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