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

Mark Michelson (JIRA) noreply at issues.asterisk.org
Wed Jan 14 14:21:34 CST 2015


     [ 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.patch

I am attaching ASTERISK-24624.patch.

With this patch, the code that normally would create a new channel when receiving an INVITE now detects if the incoming INVITE is actually a reinvite and terminates the session if that is the case.

I could not reproduce the problem exactly as you had done, but I managed to do something similar with a Digium phone. The patch clears up the issue I had, but I want to double-check that it does for you as well. Please give the patch a try and let me know if it helps.

> 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.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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