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

Sébastien M. (JIRA) noreply at issues.asterisk.org
Wed Aug 19 08:14:33 CDT 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25328?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227298#comment-227298 ] 

Sébastien M. commented on ASTERISK-25328:
-----------------------------------------

Hi,

When I read the doc, ANSWEREDTIME seems to be the bridged time : the duration where the 2 participants have spoken together.
This variable is good on Asterisk 11.

If it's not a bug, how can I obtain this information ?

Thanks

> 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: Sébastien M.
>         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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