[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}
--
This message was sent by Atlassian JIRA
(v6.2#6252)
More information about the asterisk-bugs
mailing list