[asterisk-bugs] [JIRA] Closed: (ASTERISK-19628) Crash during blind transfer with chan_unistim

[Digium Subversion (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-19628?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Digium Subversion closed ASTERISK-19628.
----------------------------------------

    Resolution: Fixed

> Crash during blind transfer with chan_unistim
> ---------------------------------------------
>
>                 Key: ASTERISK-19628
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-19628
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_unistim
>    Affects Versions: SVN
>         Environment: Debian 6.0, Asterisk-SVN-r360935, VmWare 3.5 virtual machine.
>            Reporter: Igor Olhovskiy
>            Assignee: Igor Goncharovsky
>         Attachments: backtrace1.txt, backtrace.txt, unistim.conf
>
>
> When call to i2004, trying to transfer to another SIP device, asterisk crushes. What info to be provided to detailed debug?
> All SIP abonents are on another Asterisk server. (Due to testing purposes)
> Console output is:
>     -- Starting switch on '604 at 604-0' to 602
>     -- Executing [602 at TestUNISTIM:1] Dial("USTM/604 at 604-0xa439b90", "SIP/MainAsterisk/602") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/MainAsterisk/602
>   == Using SIP RTP CoS mark 5
>     -- Executing [602 at LocalIn:1] Set("SIP/MainAsterisk-00000003", "CALLERID(name)=Main Asterisk") in new stack
>     -- Executing [602 at LocalIn:2] Dial("SIP/MainAsterisk-00000003", "USTM/602 at 602") in new stack
> [Apr  3 09:47:21] ERROR[2919][C-00000000]: astobj2.c:140 INTERNAL_OBJ: user_data is NULL
>     -- Called USTM/602 at 602
>     -- USTM/602 at 602-0xa434780 is ringing
>     -- SIP/MainAsterisk-00000002 is ringing
>     -- USTM/602 at 602-0xa434780 answered SIP/MainAsterisk-00000003
>     -- SIP/MainAsterisk-00000002 answered USTM/604 at 604-0xa439b90
>     -- Started music on hold, class 'default', on SIP/MainAsterisk-00000003
>     -- Starting switch on '602 at 602-0' to 658
>     -- Executing [658 at TestUNISTIM:1] Dial("USTM/602 at 602-0xa4a2fd8", "SIP/MainAsterisk/658") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/MainAsterisk/658
>     -- SIP/MainAsterisk-00000004 is ringing
>     -- Stopped music on hold on SIP/MainAsterisk-00000003
> [Apr  3 09:47:36] NOTICE[2907]: chan_unistim.c:2367 attempt_transfer: UNISTIM transfer: trying to masquerade SIP/MainAsterisk-00000003 into USTM/602 at 602-0xa4a2fd8
> [Apr  3 09:47:36] NOTICE[2920]: chan_unistim.c:2375 attempt_transfer: Transfer attempted with no appropriate bridged calls to transfer
> attempt_transfer failed.
> NortelStationTrunk*CLI> 
> Disconnected from Asterisk server
> Executing last minute cleanups
> Asterisk cleanly ending (0).

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