[asterisk-bugs] [JIRA] Commented: (ASTERISK-19937) Still not working - Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2, 1.8.11, 1.8.12.2

Peter Sokolov (JIRA) noreply at issues.asterisk.org
Tue Jul 17 18:13:21 CDT 2012


    [ https://issues.asterisk.org/jira/browse/ASTERISK-19937?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=194955#comment-194955 ] 

Peter Sokolov commented on ASTERISK-19937:
------------------------------------------

I was out for a few days so could not test this before. With the version 1.8.14.1, the problem is solved. CANCEL and ACK are sent to correct IPs.

It seems that now directrtpsetup=yes stopped working, which worked fine with 1.8.12.2. All calls are setup with RTP passing via Asterisk first. Before opening another issue, I will double check that all settings are still correct.

> Still not working - Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2, 1.8.11, 1.8.12.2
> -----------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-19937
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-19937
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>         Environment: Debian, Asterisk 1.8.12.2
>            Reporter: Peter Sokolov
>            Assignee: Peter Sokolov
>            Severity: Critical
>              Labels: regression
>         Attachments: annotated_SIP_debug.txt
>
>
> Not sure, this has to do with ASTERISK-19358. The visible problem is the same. That is, phones do not stop ringing after CANCEL.
> Looking into SIP debug one can see that Asterisk is sending various SIP messages (CANCEL, ACK) to the IPs that are not participating in the call at all. In the below example one can see that for example it says:
> [Edited- 6/3/12- Rusty Newton - removed debug an attached as annotated_SIP_debug.txt]

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