[asterisk-bugs] [JIRA] (ASTERISK-25563) Channel Hangup after DTMF during Playback

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25563?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228349#comment-228349 ] 

Rusty Newton commented on ASTERISK-25563:
-----------------------------------------

What dialplan is at asterisk-redirect,1,1 ?

> Channel Hangup after DTMF during Playback
> -----------------------------------------
>
>                 Key: ASTERISK-25563
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25563
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_playback, Resources/res_agi
>    Affects Versions: 13.4.0, 13.6.0
>         Environment: CentOS 7 in KVM Virtual Machine
>            Reporter: Martin Vogt
>         Attachments: debug9_25563.txt
>
>
> We have an Adhearsion based AGI application.
> In Adheasion we use the function menu that starts the playback function in asterisk.
> If we send a dtmf to the channel during the playback, the normal behavior should be that asterisk stops the playback an continues the agi application. but we get a hangup on the channel.
> extensions.conf:
> {noformat}
> [from_sip]
> exten => 777,1,NoOp(Extension: ${EXTEN})
> exten => 777,2,Set(AGI_HOST=127.0.0.1)
> exten => 777,3,Goto(menu_test,${EXTEN},1) 
> [menu_test]
> exten => _777,1,AGI(agi:async)  
> {noformat}
> adhearsion dialplan:
> {noformat}
> class TestController < Adhearsion::CallController
>   def run
>     answer
>     # menu %w(file://vm-press file://vm-first file://vm-for file://vm-password), :limit => 1, :timeout => 5.seconds do
>     menu %w(file://vm-press file://vm-first file://vm-for file://vm-password), :terminator => '#' do
>       match [1, 2, 3] do |dial|
>         logger.info "Caller pressed #{dial}"
>       end
>     end
>     play 'file://vm-pls-try-again'
>     #####################################
>     # this menu is never audible
>     #####################################
>     menu %w(file://vm-press file://vm-first file://vm-for file://vm-password), :limit => 1 do
>       match [1, 2, 3] do |dial|
>         logger.info "Caller pressed #{dial}"
>       end
>     end
>     hangup
>   end
> end
> {noformat}
> {noformat}
> SIP/111 dials 777 ---> 777 at from_sip ---> 777 at menu_test ---> AGI ---> menu (adhearsion) ---> Playback(asterisk) ---> dtmf to channel ---> hangup
> {noformat}
> We did a tcpdump and found out that adhearsion is not sending the hangup request to asterisk. So we think asterisk clears the call on its own.
> We have already opened an issue on the adhearsion bug tracker: https://github.com/adhearsion/adhearsion-asr/issues/29 
> The Adhearsion developement team also thinks that asterisk clears the call by itself.
> debug logs follow...



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