[asterisk-bugs] [JIRA] (ASTERISK-24620) AGI GET VARIABLE ANSWEREDTIME gives ZERO on callback bridge using app

Rusty Newton (JIRA) noreply at issues.asterisk.org
Mon Dec 15 09:53:29 CST 2014


    [ https://issues.asterisk.org/jira/browse/ASTERISK-24620?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224006#comment-224006 ] 

Rusty Newton edited comment on ASTERISK-24620 at 12/15/14 9:52 AM:
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We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


Specifically make sure to get all of the logger channel types, and make sure the DEBUG messages appear in your log before attaching it to the issue.

Press Send Back or Enter Feedback to send the issue back to us.


was (Author: rnewton):
We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


Specifically make sure to get all of the logger channel types, and make sure the DEBUG messages appear in your log before attaching it to the issue.

> AGI GET VARIABLE ANSWEREDTIME gives ZERO on callback bridge using app
> ---------------------------------------------------------------------
>
>                 Key: ASTERISK-24620
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24620
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_dial
>    Affects Versions: 13.0.2
>         Environment: CentOS 6.5 64bit
>            Reporter: Sid Mason
>         Attachments: debug.txt
>
>
> Below is the log of Asterisk Console during Callback request by 6012XXXXXXX
> a) when call SIP/sipprovider/6012XXXXXXX is connected
> b) Asterisk is dialing another sip channel SIP/ims_peer/60327XXXXXX
> c) call duriation on SIP/ims_peer/60327XXXXXX is 00:00:12
> d) GET VARIABLE ANSWEREDTIME report Wrong : 0
> e) GET VARIABLE DIALEDTIME report Wrong : 28 
> [Edit by Rusty - Remove inline debug - logs don't go in the description field. In the future, please use attachments for logs.]



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