[asterisk-bugs] [JIRA] (ASTERISK-27446) No 'h' extension run when call transferred by caller completes

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27446?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=240281#comment-240281 ] 

Asterisk Team commented on ASTERISK-27446:
------------------------------------------

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].

> No 'h' extension run when call transferred by caller completes
> --------------------------------------------------------------
>
>                 Key: ASTERISK-27446
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27446
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.17.2
>         Environment: CentOS 7.2 64 bit.
>            Reporter: David Cunningham
>
> When a caller does a REFER transfer to connect two calls, the 'h' extension is run immediately, and no 'h' extension is run when the transferred call actually hangs up. For example:
> 1000105 calls 1000103
> 1000105 calls 1000102
> 1000105 transfers one call to the other, and drops out of the call
> 1000103 and 1000102 talk
> 1000103 or 1000102 hangs up and the call completes, but no 'h' extension is run
> This means that our code on the Asterisk server can't correctly bill for the transferred call. This issue didn't happen on Asterisk 11 and prior.
> I've attached the dial plan, and an Asterisk console trace in case it's useful. Important points are:
> Line 142: Dial plan logs start of call 1511749212.44
> Line 668: Dial plan logs start of call 1511749221.46
> Line 950: REFER happens
> Line 1010: Dial plan logs end of call 1511749212.44
> Line 1013: Dial plan logs end of call 1511749221.46
> Line 1240: Bridge torn down after transferred call ends, but no end of call is logged because 'h' extension is not run
> Please let us know if anything else would be useful.



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