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Summary:ASTERISK-14579: Asterisk crushes if I hangup call immediately after I make it from google talk to my Asterisk
Reporter:Gennady G. Marchenko (zgen)Labels:
Date Opened:2009-08-02 09:06:24Date Closed:2011-06-07 14:01:00
Priority:CriticalRegression?No
Status:Closed/CompleteComponents:Channels/chan_gtalk
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:OS: Linux/Debian 5
Kernel: 2.6.26-1-686 #1 SMP

I make call from google talk to my asterisk and immediately hangup the call. The result is Asterisk crushed. If I do this not so fast, all works fine.

I reproduced this problem three times.


****** ADDITIONAL INFORMATION ******

There is some debug jabber info:

ast*CLI>
JABBER: asterisk INCOMING: <iq to="call_to(asterisk)@zgen.ru/Talk2AB958BE" type="set" id="137" from="call_from@gmail.com/Talk.v104B7F43F50"><session type="initiate" id="890611275" initiator="call_from@gmail.com/Talk.v104B7F43F50" xmlns="http://www.google.com/session"><description xml:lang="en" xmlns="http://www.google.com/session/phone"><payload-type id="103" name="ISAC" clockrate="16000"/><payload-type id="97" name="IPCMWB" clockrate="16000" bitrate="80000"/><payload-type id="99" name="speex" clockrate="16000" bitrate="22000"/><payload-type id="4" name="G723" clockrate="8000" bitrate="6300"/><payload-type id="98" name="speex" clockrate="8000" bitrate="11000"/><payload-type id="100" name="EG711U" clockrate="8000" bitrate="64000"/><payload-type id="101" name="EG711A" clockrate="8000" bitrate="64000"/><payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"/><payload-type id="8" name="PCMA" clockrate="8000" bitrate="64000"/><payload-type id="13" name="CN" clockrate="8000"/><payload-type id="102" name="iLBC"

JABBER: asterisk INCOMING: clockrate="8000" bitrate="13300"/><payload-type id="106" name="telephone-event" clockrate="8000"/></description><transport xmlns="http://www.google.com/transport/p2p"/></session></iq>
ast*CLI>
JABBER: asterisk OUTGOING: <iq type='result' from='call_to(asterisk)@zgen.ru/Talk2AB958BE' to='call_from@gmail.com/Talk.v104B7F43F50' id='137'/>

JABBER: asterisk OUTGOING: <message type='chat' to='call_from@gmail.com' from='call_to(asterisk)@zgen.ru/Talk2AB958BE'><body>You are calling my home Asterisk PBX. Please wait...</body></message>

JABBER: asterisk OUTGOING: <iq from='call_to(asterisk)@zgen.ru/Talk2AB958BE' to='call_from@gmail.com/Talk.v104B7F43F50' type='set' id='aaaad'><session type='transport-accept' id='890611275' initiator='call_from@gmail.com/Talk.v104B7F43F50' xmlns='http://www.google.com/session'><transport xmlns='http://www.google.com/transport/p2p'/></session></iq>

JABBER: asterisk OUTGOING: <iq from='call_to(asterisk)@zgen.ru/Talk2AB958BE' to='call_from@gmail.com/Talk.v104B7F43F50' type='set' id='aaaae'><session type='transport-info' id='890611275' initiator='call_from@gmail.com/Talk.v104B7F43F50' xmlns='http://www.google.com/session'><transport xmlns='http://www.google.com/transport/p2p'><candidate name='rtp' address='87.245.189.162' port='32678' username='401618355825ce59' password='65dd6b1b2aeca198' preference='1.00' protocol='udp' type='local' network='0' generation='0'/></transport></session></iq>

JABBER: asterisk OUTGOING: <message type='chat' to='call_from@gmail.com' from='call_to(asterisk)@zgen.ru/Talk2AB958BE'><body>2009-08-02-17:16:59-CID-7002-ID1249219019.0-.wav</body></message>
[Aug  2 17:16:59] WARNING[14190]: res_jabber.c:1573 aji_recv_loop: JABBER: socket read error
ast*CLI>
JABBER: asterisk OUTGOING: <?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='zgen.ru' version='1.0'>
ast*CLI>
JABBER: asterisk INCOMING: <stream:stream from="zgen.ru" id="8EB4985A6B0EC7EF" version="1.0" xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client">
ast*CLI>
JABBER: asterisk OUTGOING: <starttls xmlns='urn:ietf:params:xml:ns:xmpp-tls'/>
[Aug  2 17:17:00] NOTICE[14224]: chan_gtalk.c:1514 gtalk_indicate: Don't know how to indicate condition '3'
[Aug  2 17:17:00] WARNING[14224]: translate.c:175 framein: no samples for ulawtolin
ast*CLI>
JABBER: asterisk INCOMING: <stream:features><starttls xmlns="urn:ietf:params:xml:ns:xmpp-tls"><required/></starttls><mechanisms xmlns="urn:ietf:params:xml:ns:xmpp-sasl"><mechanism>X-GOOGLE-TOKEN</mechanism></mechanisms></stream:features><proceed xmlns="urn:ietf:params:xml:ns:xmpp-tls"/>
ast*CLI>
JABBER: asterisk OUTGOING: <?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='zgen.ru' version='1.0'>
ast*CLI>
JABBER: asterisk INCOMING: <stream:stream from="zgen.ru" id="B14F2413B75891BD" version="1.0" xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client">
ast*CLI>
JABBER: asterisk INCOMING: <stream:features><mechanisms xmlns="urn:ietf:params:xml:ns:xmpp-sasl"><mechanism>PLAIN</mechanism><mechanism>X-GOOGLE-TOKEN</mechanism></mechanisms></stream:features>
ast*CLI>
JABBER: asterisk OUTGOING: <auth xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='PLAIN'>AHNlbGYAYXJtdHdqYjk=</auth>
ast*CLI>
JABBER: asterisk INCOMING: <success xmlns="urn:ietf:params:xml:ns:xmpp-sasl"/>
ast*CLI>
JABBER: asterisk OUTGOING: <?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='zgen.ru' version='1.0'>
ast*CLI>
JABBER: asterisk INCOMING: <stream:stream from="zgen.ru" id="219178EAFE9B9AB5" version="1.0" xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client">
ast*CLI>
JABBER: asterisk INCOMING: <stream:features><bind xmlns="urn:ietf:params:xml:ns:xmpp-bind"/><session xmlns="urn:ietf:params:xml:ns:xmpp-session"/></stream:features>
ast*CLI>
JABBER: asterisk OUTGOING: <iq type='set' id='aaaaf'><bind xmlns='urn:ietf:params:xml:ns:xmpp-bind'><resource>Talk2AB958BE</resource></bind></iq>
ast*CLI>
JABBER: asterisk OUTGOING: <iq type='set' id='auth'><session xmlns='urn:ietf:params:xml:ns:xmpp-session'/></iq>
ast*CLI>
JABBER: asterisk INCOMING: <iq id="aaaaf" type="result"><bind xmlns="urn:ietf:params:xml:ns:xmpp-bind"><jid>call_to(asterisk)@zgen.ru/Talk2AB958BE</jid></bind></iq>
ast*CLI>
JABBER: asterisk OUTGOING: <presence from='call_to(asterisk)@zgen.ru/Talk2AB958BE'><status>.This is an Asterisk server.</status><c node='http://www.asterisk.org/xmpp/client/caps' ver='asterisk-xmpp' ext='voice-v1' xmlns='http://jabber.org/protocol/caps'/></presence>
ast*CLI>
JABBER: asterisk OUTGOING: <iq type='get' id='roster'><query xmlns='jabber:iq:roster'/></iq>
ast*CLI>
JABBER: asterisk INCOMING:
ast*CLI>
JABBER: asterisk INCOMING: <iq type="result" id="auth"/>
ast*CLI>
JABBER: asterisk INCOMING: <iq to="call_to(asterisk)@zgen.ru/Talk2AB958BE" id="roster" type="result"><query xmlns="jabber:iq:roster"><item jid="call_from@gmail.com" subscription="both"/><item jid="gennady.archenko@gmail.com" subscription="none" ask="subscribe"/></query></iq>
ast*CLI>
JABBER: asterisk INCOMING:
ast*CLI>
JABBER: asterisk INCOMING: <presence from="call_from@gmail.com/Talk.v104B7F43F50" to="call_to(asterisk)@zgen.ru/Talk2AB958BE"><priority>24</priority><c node="http://www.google.com/xmpp/client/caps" ver="1.0.0.104" ext="share-v1 voice-v1" xmlns="http://jabber.org/protocol/caps"/><x stamp="20090802T13:08:03" xmlns="jabber:x:delay"/><status/><x xmlns="vcard-temp:x:update"><photo/></x></presence>
[Aug  2 17:17:11] NOTICE[14224]: chan_gtalk.c:1514 gtalk_indicate: Don't know how to indicate condition '-1'
ast*CLI>
JABBER: asterisk OUTGOING: <iq type='set' to='call_from@gmail.com/Talk.v104B7F43F50' from='call_to(asterisk)@zgen.ru/Talk2AB958BE' id='aaaah'><session xmlns='http://www.google.com/session' type='accept' initiator='call_from@gmail.com/Talk.v104B7F43F50' id='890611275'><description xmlns='http://www.google.com/session/phone' xml:lang='en'><payload-type/></description><transport xmlns='http://www.google.com/transport/p2p'/></session></iq>
[Aug  2 17:17:11] NOTICE[14224]: chan_gtalk.c:1514 gtalk_indicate: Don't know how to indicate condition '-1'
[Aug  2 17:17:11] NOTICE[14224]: chan_gtalk.c:1514 gtalk_indicate: Don't know how to indicate condition '20'
ast*CLI>
JABBER: asterisk INCOMING: <iq to="call_to(asterisk)@zgen.ru/Talk2AB958BE" id="aaaah" type="error" from="call_from@gmail.com/Talk.v104B7F43F50"><session type="accept" initiator="call_from@gmail.com/Talk.v104B7F43F50" id="890611275" xmlns="http://www.google.com/session"><description xml:lang="en" xmlns="http://www.google.com/session/phone"><payload-type/></description><transport xmlns="http://www.google.com/transport/p2p"/></session><error type="modify"><sta:bad-request xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"/><sta:text xml:lang="en" xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas">unknown session</sta:text></error></iq>
ast*CLI>
Disconnected from Asterisk server

gdb doesn't show something interesting:

Loaded symbols for /usr/lib/asterisk/modules/func_timeout.so
Core was generated by `asterisk -vvvvg'.
Program terminated with signal 11, Segmentation fault.
[New process 14327]
[New process 14376]
[New process 14375]
[New process 14366]
[New process 14365]
[New process 14364]
[New process 14361]
[New process 14360]
[New process 14359]
[New process 14358]
[New process 14357]
[New process 14356]
[New process 14355]
[New process 14354]
[New process 14353]
[New process 14352]
[New process 14351]
[New process 14350]
[New process 14349]
[New process 14348]
[New process 14347]
[New process 14346]
[New process 14345]
[New process 14344]
[New process 14343]
[New process 14342]
[New process 14341]
[New process 14340]
[New process 14329]
[New process 14328]
[New process 14323]
[New process 14326]
[New process 14325]
[New process 14324]
#0  0xb7d48358 in ?? () from /lib/i686/cmov/libc.so.6



Comments:By: Gennady G. Marchenko (zgen) 2009-08-02 13:15:24

call was cancelled but my phone is ringing, when I take it, asterisk cruches
By: Leif Madsen (lmadsen) 2009-09-15 12:41:48

Do you have DONT_OPTIMIZE in the Compiler Flags of menuselect enabled? If not, please do that, then reinstall Asterisk and reproduce this issue. You should be able to get a useful backtrace that way.

Once you have done that, please attach it here. You may also follow the instructions in doc/backtrace.txt of your Asterisk source.
By: Leif Madsen (lmadsen) 2009-09-30 09:55:34

I'm closing this issue as it has been waiting on feedback for more than 2 weeks. If the reporter has additional information that can be submitted, than they are welcome to reopen the issue. Thanks!