[asterisk-bugs] [JIRA] (ASTERISK-22703) crash: hanging up channels in a softmix bridge

Matt DiMeo (JIRA) noreply at issues.asterisk.org
Mon Oct 21 17:24:03 CDT 2013


    [ https://issues.asterisk.org/jira/browse/ASTERISK-22703?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=211106#comment-211106 ] 

Matt DiMeo edited comment on ASTERISK-22703 at 10/21/13 5:23 PM:
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I see possibly the same crash all the time on the current trunk.


dial into my stasis app from my sip phone

Answer with stasis:

calling http.request with: {"path":"/ari/channels/1381953659.0/answer","method":"POST","hostname":"127.0.0.1","port":8088,"auth":"x:x"}

Start music on hold:

calling http.request with: {"path":"/ari/channels/1381953659.0/mohstart?mohClass=","method":"POST","urlParams":{"mohClass":""},"hostname":"127.0.0.1","port":8088,"auth":"x:x"}


My app dials to another sip phone:

calling http.request with: {"path":"/ari/channels?endpoint=SIP%2F1001&context=&extension=&priority=&app=queue&appArgs=agent%2C2%2C1381953657375%2C801&callerId=my%20first%20queue!%3C801%3E&timeout=30","method":"POST","urlParams":{"endpoint":"SIP/1001","app":"queue","appArgs":"agent,2,1381953657375,801","callerId":"my first queue!<801>","timeout":30},"hostname":"127.0.0.1","port":8088,"auth":"x:x"}

App creates a bridge:

calling http.request with: {"path":"/ari/bridges?type=","method":"POST","urlParams":{"type":""},"hostname":"127.0.0.1","port":8088,"auth":"x:x"}


App bridges the two channels:

calling http.request with: {"path":"/ari/bridges/9b2f06bd-e21c-4310-9364-058e82871476/addChannel?channel=1381953659.0%2C1381953659.1","method":"POST","urlParams":{"channel":"1381953659.0,1381953659.1"},"hostname":"127.0.0.1","port":8088,"auth":"x:x"}


Then I hang up the first phone that dialed into the app.

Then I hang up the second phone, and asterisk crashes.

                
      was (Author: mdimeo):
    I see possibly the same crash all the time on the current trunk.


dial into my stasis app from my sip phone

Answer with stasis:

calling http.request with: {"path":"/ari/channels/1381953659.0/answer","method":"POST","hostname":"127.0.0.1","port":8088,"auth":"bmd:fourloop"}

Start music on hold:

calling http.request with: {"path":"/ari/channels/1381953659.0/mohstart?mohClass=","method":"POST","urlParams":{"mohClass":""},"hostname":"127.0.0.1","port":8088,"auth":"bmd:fourloop"}


My app dials to another sip phone:

calling http.request with: {"path":"/ari/channels?endpoint=SIP%2F1001&context=&extension=&priority=&app=queue&appArgs=agent%2C2%2C1381953657375%2C801&callerId=my%20first%20queue!%3C801%3E&timeout=30","method":"POST","urlParams":{"endpoint":"SIP/1001","app":"queue","appArgs":"agent,2,1381953657375,801","callerId":"my first queue!<801>","timeout":30},"hostname":"127.0.0.1","port":8088,"auth":"bmd:fourloop"}

App creates a bridge:

calling http.request with: {"path":"/ari/bridges?type=","method":"POST","urlParams":{"type":""},"hostname":"127.0.0.1","port":8088,"auth":"bmd:fourloop"}


App bridges the two channels:

calling http.request with: {"path":"/ari/bridges/9b2f06bd-e21c-4310-9364-058e82871476/addChannel?channel=1381953659.0%2C1381953659.1","method":"POST","urlParams":{"channel":"1381953659.0,1381953659.1"},"hostname":"127.0.0.1","port":8088,"auth":"bmd:fourloop"}


Then I hang up the first phone that dialed into the app.

Then I hang up the second phone, and asterisk crashes.

                  
> crash: hanging up channels in a softmix bridge
> ----------------------------------------------
>
>                 Key: ASTERISK-22703
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22703
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp, Bridges/bridge_softmix
>    Affects Versions: 12.0.0-beta1
>         Environment: Asterisk trunk r400862
>            Reporter: John Bigelow
>         Attachments: backtrace2.txt, backtrace.txt, full2.txt
>
>
> I did the following:
> * put two SIP channels in a bridge via ARI (native rtp)
> * ensured audio sounded fine
> * added a third SIP channel to the bridge (softmix)
> * ensured the audio sounded fine
> * Hung up channel last channel that was added to the bridge by hanging up phone
> * Asterisk crashed
> I've attached the backtrace and Asterisk full log.

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