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Summary:ASTERISK-25819: AMI hangup "cause" value ignored or overridden when channel is hungup during process of origination
Reporter:Dialapplet (Dialapplet)Labels:
Date Opened:2016-02-29 05:24:05.000-0600Date Closed:2018-01-02 08:44:22.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/General
Versions:11.21.2 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Attachments:
Description:When a channel is hung up from asterisk manager during a call and a cause code is specified all works fine. But, when I try to hung up a channel that is in process of origination, it is hung up with cause 19, ignoring the cause I put on the action.

For example:
I do this orginate:
{quote}
Action: Originate
Async: true
Channel: SIP/100
...
{quote}

The channel is created:
{quote}
Event: Newchannel
Channel: SIP/100-00000005
Uniqueid: 1456738121.6
...
{quote}

I do the hang up action with cause code 17 (When the phone is still ringing, just to reject the call):
{quote}
Action: Hangup
Cause: 17
Channel: SIP/100-00000005
ActionID: Hangup-100
{quote}

I receive the response:
{quote}
ActionID: Hangup-100
Response: Success
Message: Channel Hungup
{quote}

But the hangup event doesn't contain cause code 17:
{quote}
Event: Hangup
Cause: 19
Channel: SIP/100-00000005
Uniqueid: 1456738121.6
...
{quote}
Comments:By: Asterisk Team (asteriskteam) 2016-02-29 05:24:06.504-0600

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].
By: Dialapplet (Dialapplet) 2016-02-29 05:55:43.677-0600

I edited channel.c to output hangup cause on line 2841 and the output is the following:

{quote}
DEBUG\[7955\]: channel.c:628 ast_channel_softhangup_withcause_locked: Setting hangupcause of channel SIP/100-00000000 to 17 (is 16 now)
DEBUG\[7955\]: channel.c:2662 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/100-00000000'
DEBUG\[7980\]\[C-00000000\]: channel.c:2841 ast_hangup: Hanging up channel 'SIP/100-00000000' with cause '19'
{quote}

The hangup cause is overwritten in channel.cpp:5854
{quote}
if (AST_CONTROL_RINGING == last_subclass) {
   ast_channel_hangupcause_set(chan, AST_CAUSE_NO_ANSWER);
}
{quote}
By: Rusty Newton (rnewton) 2016-03-01 13:45:33.313-0600

Thanks and have you looked or tested to see if this issue is present in 13?
By: Dialapplet (Dialapplet) 2016-03-02 02:33:02.000-0600

No, I didn't. We are working on a integration with version 13 but currently our software it's incompatible with the new changes since version 12 involving the AMI events
By: Joshua C. Colp (jcolp) 2017-12-18 08:48:30.576-0600

Asterisk 11 is no longer supported so confirmation of this against 13 would be good. Things have changed there so it may no longer be applicable.
By: Asterisk Team (asteriskteam) 2018-01-02 08:44:22.661-0600

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines