[asterisk-bugs] [JIRA] (ASTERISK-24681) Asterisk restarts during attended transfer if on-hold peer is disconnected before the ringing tones are heard

Rusty Newton (JIRA) noreply at issues.asterisk.org
Thu Jan 29 18:44:35 CST 2015


     [ https://issues.asterisk.org/jira/browse/ASTERISK-24681?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-24681:
------------------------------------

    Assignee: Max Man  (was: Matt Jordan)
      Status: Waiting for Feedback  (was: Triage)

Max, we will need to close this issue out if you can't provide a backtrace. We don't have your exact configuration and therefore cannot reproduce this issue.

 Without a backtrace we have nothing to go on...

> Asterisk restarts during attended transfer if on-hold peer is disconnected before the ringing tones are heard
> -------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24681
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24681
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.0.1
>         Environment: FREEPBX Distro 6.12.65-22.
> faxdetect=no
> context=from-sip-external
> callerid=Unknown
> notifyringing=yes
> notifyhold=yes
> tos_sip=cs3
> tos_audio=ef
> tos_video=af41
> alwaysauthreject=yes
> useragent=FPBX-12.0.13(13.0.1)
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> allow=g726
> allow=g722
> progressinband=no
> callevents=no
> language=ru
> rtpend=20000
> rtpstart=10000
> bindport=5060
> jbenable=no
> minexpiry=60
> defaultexpiry=120
> registerattempts=0
> maxexpiry=3600
> notifyhold=yes
> notifyringing=yes
> srvlookup=no
> allowguest=no
> g726nonstandard=no
> videosupport=no
> maxcallbitrate=384
> canreinvite=no
> rtptimeout=30
> rtpholdtimeout=300
> rtpkeepalive=0
> checkmwi=10
> registertimeout=20
> nat=yes 
>            Reporter: Max Man
>            Assignee: Max Man
>
> Making internal call from (1) ext to (2) another, answering the call on (2), pushing ** (which is feature code for attended transfer), dialing the ext number (f.e. 222) on (2) and at the same time (before the ringing tones are heard) hanging down the (1) phone, after that getting the following in the console:
>  
> ERROR[2969]: astobj2.c:116 INTERNAL_OBJ: user_data is NULL
> Disconnected from Asterisk server
> Asterisk cleanly ending (0).
> Executing last minute cleanups
> Asterisk process is restarted... seems like this bug can cause all active calls to disconnect.



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