[asterisk-bugs] [JIRA] (ASTERISK-24624) Transfer to invalid extension results in hung channel.

[Mark Michelson (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24624?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Mark Michelson updated ASTERISK-24624:
--------------------------------------

    Attachment: ASTERISK-24624-13.1.0.patch

I have deleted the previous 13.1.0 version of the patch and added a new one here. I did not realize that ast_sip_session_terminate() was not defined in 13.1.0, so I have replaced the code with an in-line generation of the BYE request. This will behave differently than the 13 branch version. The 13 branch version will wait until the ACK is received to send the BYE since it has logic baked in to wait until INVITE transactions have cleared to send the BYE.

The 13.1.0 version of this patch will send the BYE immediately instead of waiting for an ACK. This may result in bad behavior on your devices, though it should be fine.

> Transfer to invalid extension results in hung channel.
> ------------------------------------------------------
>
>                 Key: ASTERISK-24624
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24624
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.0.0, 13.0.1, 13.0.2, 13.1.0
>            Reporter: Zane Conkle
>            Assignee: Mark Michelson
>         Attachments: ASTERISK-24624-13.1.0.patch, ASTERISK-24624.patch
>
>
> When attempting to blind transfer a call to an invalid extension or resource the channel will hang in the "Ring" state.
> I was able to duplicate this by blind transferring a call to a mailbox that does not exist. Below is the CLI output along with the stuck channel.
> {code}    -- Executing [s200-DEFAULT at default:1] Dial("SIP/carrier1-00000002", "PJSIP/200_default,10")
>     -- Called PJSIP/200_default
>     -- PJSIP/200_default-00000002 is ringing
>     -- PJSIP/200_default-00000002 answered SIP/carrier1-00000002
>     -- Channel SIP/carrier1-00000002 joined 'simple_bridge' basic-bridge <44f9110a-2508-4e62-acb4-75beca76ce31>
>     -- Channel PJSIP/200_default-00000002 joined 'simple_bridge' basic-bridge <44f9110a-2508-4e62-acb4-75beca76ce31>
>     -- Started music on hold, class 'default', on channel 'SIP/carrier1-00000002'
>     -- Stopped music on hold on SIP/carrier1-00000002
>     -- Channel SIP/carrier1-00000002 left 'simple_bridge' basic-bridge <44f9110a-2508-4e62-acb4-75beca76ce31>
>     -- Channel PJSIP/200_default-00000002 left 'simple_bridge' basic-bridge <44f9110a-2508-4e62-acb4-75beca76ce31>
>     -- Executing [1234 at default:1] VoiceMail("SIP/carrier1-00000002", "5678 at default") in new stack
> [Dec 16 10:29:58] WARNING[3204][C-00000002]: app_voicemail.c:6467 leave_voicemail: No entry in voicemail config file for '5678'
>     -- Auto fallthrough, channel 'SIP/carrier1-00000002' status is 'ANSWER'
> [root at pbx ~]# asterisk -rx "core show channels verbose" 
> Channel              Context              Extension        Prio State   Application  Data                      CallerID        Duration Accountcode PeerAccount BridgeID            
> PJSIP/200_default    default              s                   1 Ring    (None)       (Empty)                   200             00:27:57 default 
> {code}



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