[asterisk-bugs] [JIRA] (ASTERISK-25563) Channel Hangup after DTMF during Playback
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Mon Nov 16 07:18:33 CST 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-25563?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228319#comment-228319 ]
Asterisk Team commented on ASTERISK-25563:
------------------------------------------
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> 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
>
> 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:
> [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)
> adhearsion dialplan:
> 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
> SIP/111 dials 777 ---> 777 at from_sip ---> 777 at menu_test ---> AGI ---> menu (adhearsion) ---> Playback(asterisk) ---> dtmf to channel ---> hangup
> 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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