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

[Andrey Petrov (JIRA)]

res_xmpp.c: JABBER: socket read error, afterwards outgoing connections never get answered
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                 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:

[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

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:
[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


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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