[asterisk-bugs] [JIRA] (ASTERISK-25328) Wrong ANSWEREDTIME after Dial app execution
Sébastien M. (JIRA)
noreply at issues.asterisk.org
Thu Aug 20 04:25:33 CDT 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-25328?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227314#comment-227314 ]
Sébastien M. commented on ASTERISK-25328:
-----------------------------------------
Thanks for your response.
I need to answer the call because I have an IVR before the dial command.
A simple example :
{quote}
exten => _017013xxxx,1,Answer()
exten => _017013xxxx,n,Playback(welcome)
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}
> 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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