[asterisk-bugs] [JIRA] (ASTERISK-24633) Asterisk Replies : Call leg/transaction does not exist

[Sid Mason (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24633?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Sid Mason updated ASTERISK-24633:
---------------------------------

          Description: 
It seems asterisk erases the information about the ongoing call through Chan_SIP 

This happens when :

a) 20 or AMI listener is attached to asterisk on port 5038 and they all have filter as below:
b) AMI has filter of the UNIQUE ID of the channels and others:

'!Event: AGIExec'
'!Event: VarSet'
'!Event: RTCPSent'
'!Event: RTCPReceived'
'!Event: VarSet'

c) the call durations are usually above 10 minutes

d) the bridge is doing unlink before the hang-up and somehow asterisk must hold the information for SIP Dialog right before the hang-up

I tried asterisk 11.15 and the same result was there.

At this point ANSWEREDTIME is reported as NULL as well

  was:It seems asterisk erases the information about the ongoing call through Chan_SIP 

    Affects Version/s: 11.15.0

> Asterisk Replies : Call leg/transaction does not exist 
> -------------------------------------------------------
>
>                 Key: ASTERISK-24633
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24633
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.15.0, 13.1.0
>         Environment: CentOS 6.5 64bit
>            Reporter: Sid Mason
>            Assignee: Sid Mason
>            Severity: Critical
>         Attachments: DEBUG_LOG Asterisk 11.15.txt, DEBUG_LOG.txt
>
>
> It seems asterisk erases the information about the ongoing call through Chan_SIP 
> This happens when :
> a) 20 or AMI listener is attached to asterisk on port 5038 and they all have filter as below:
> b) AMI has filter of the UNIQUE ID of the channels and others:
> '!Event: AGIExec'
> '!Event: VarSet'
> '!Event: RTCPSent'
> '!Event: RTCPReceived'
> '!Event: VarSet'
> c) the call durations are usually above 10 minutes
> d) the bridge is doing unlink before the hang-up and somehow asterisk must hold the information for SIP Dialog right before the hang-up
> I tried asterisk 11.15 and the same result was there.
> At this point ANSWEREDTIME is reported as NULL as well



--
This message was sent by Atlassian JIRA
(v6.2#6252)