[asterisk-bugs] [JIRA] Commented: (ASTERISK-20453) res_xmpp.c: JABBER: socket read error, afterwards outgoing connections never get answered

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20453?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=197555#comment-197555 ] 

Matt Jordan commented on ASTERISK-20453:
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We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Please also provide your xmpp.conf and motif.conf configuration files.

> res_xmpp.c: JABBER: socket read error, afterwards outgoing connections never get answered
> -----------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-20453
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20453
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_motif, Resources/res_xmpp
>    Affects Versions: 11.0.0-beta1
>         Environment: Linux lucid 32bit running under openvz
>            Reporter: Andrey Petrov
>            Severity: Critical
>
> Hello,
> I have been using asterisk with google voice already for a 1+ years.
> Recently I decided to switch to new asterisk11beta1, because of all new goodies coming with new Motif.
> I switched about 2 weeks ago and everything worked pretty well.
> However just a few days ago, XMPP connections seem to start ending abruptly:
> {noformat}
> [Sep 18 04:47:56] WARNING[8510] res_xmpp.c: JABBER: socket read error
> [Sep 18 07:17:00] WARNING[8510] res_xmpp.c: JABBER: socket read error
> [Sep 18 07:20:22] WARNING[8510] res_xmpp.c: JABBER: socket read error
> [Sep 18 07:21:12] WARNING[8510] res_xmpp.c: JABBER: socket read error
> {noformat}
> Interestingly enough, this error first started occuring pretty seldom, perhaps 1 a day. So in the begginning I discarded it as mundane tcp error. However, the frequency somehow increased and now it happens every other minute. It also happened to me during actual call several times. I tried setting keepalive=yes in xmpp.conf but to no avail. I tried different transports for Motif and it didnt help either.
> If I connected with asterisk -r,
> "xmpp show connections" seems to show connection is alive. And it is alive, as I am receiving status notifications.
> However, outgoing calls (and perhaps incoming, but I am not sure) placed through Motif never get answered.
> I am getting something like this in the log when placing a call:
> {noformat}
> [Sep 18 08:05:54] VERBOSE[8542][C-00000000] netsock2.c:   == Using SIP RTP CoS mark 5
> [Sep 18 08:05:54] VERBOSE[9012][C-00000000] pbx.c:     -- Executing [9093900003 at outbound:1] MixMonitor("SIP/android-00000000", "to-9093900003-Tue Sep 18 08:05:54 2012.wav,b") in new stack
> [Sep 18 08:05:54] VERBOSE[9013][C-00000000] app_mixmonitor.c:   == Begin MixMonitor Recording SIP/android-00000000
> [Sep 18 08:05:54] VERBOSE[9012][C-00000000] pbx.c:     -- Executing [9093900003 at outbound:2] Dial("SIP/android-00000000", "Motif/google/9093900003 at voice.google.com,,Tr") in new stack
> [Sep 18 08:05:54] VERBOSE[9012][C-00000000] app_dial.c:     -- Called Motif/google/9093900003 at voice.google.com
> [Sep 18 08:05:55] VERBOSE[9012][C-00000000] app_dial.c:     -- Motif/9093900003 at voice.google.com-a3a3 is proceeding passing it to SIP/android-00000000
> [Sep 18 08:06:10] VERBOSE[9012][C-00000000] pbx.c:   == Spawn extension (outbound, 9093900003, 2) exited non-zero on 'SIP/android-00000000'
> [Sep 18 08:06:11] VERBOSE[9013][C-00000000] app_mixmonitor.c:   == MixMonitor close filestream (mixed)
> [Sep 18 08:06:11] VERBOSE[9013][C-00000000] app_mixmonitor.c:   == End MixMonitor Recording SIP/android-00000000
> {noformat}
> This is a critical issue to me, because it forced me to downgrade back to 1.8.
> Additional information:
> I tried trunk, branches/11, but I can not confirm or deny if it is reproducible or not, because with these versions once call is established the voice is never heard both ways for unknown reason.

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