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Summary:ASTERISK-07936: Segmentation fault when calling Dial with a Jingle channel
Reporter:Morten Isaksen (misaksen)Labels:
Date Opened:2006-10-16 13:37:02Date Closed:2006-10-19 09:57:25
Priority:CriticalRegression?No
Status:Closed/CompleteComponents:Channels/chan_jingle
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) gtalk_diff1.diff
Description:Segmentation fault when executing

Executing [1101@sipclients:1] Dial("SIP/1014-096b2178", "Jingle/asterisk/misaksen@gmail.com") in new stack

(gdb) bt
#0  0x00cb18d3 in jingle_alloc (client=0x9caa4d0, from=0x296d839 "misaksen@gmail.com", sid=0x0) at chan_jingle.c:711
#1  0x00cb3468 in jingle_request (type=0x296d920 "Jingle", format=8, data=0x9caa4d0, cause=0x296db18) at chan_jingle.c:1384
#2  0x0807b470 in ast_request (type=0x296d920 "Jingle", format=8, data=0x0, cause=0x0) at channel.c:2824
#3  0x00d2859e in dial_exec_full (chan=0x9cceb88, data=0x296d920, peerflags=0x296e918) at app_dial.c:1074
#4  0x00d2cb1e in dial_exec (chan=0x0, data=0x6) at app_dial.c:1645
ASTERISK-1  0x080c1a16 in pbx_extension_helper (c=0x9cceb88, con=0x0, context=0x9cced08 "sipclients", exten=0x9cced58 "1101", priority=1, label=0x0, callerid=0x9c3f560 "1014", action=164080404)
   at pbx.c:505
ASTERISK-2  0x080c2d2f in __ast_pbx_run (c=0x9cceb88) at pbx.c:2245
ASTERISK-3  0x080c4c1e in pbx_thread (data=0x0) at pbx.c:2556
ASTERISK-4  0x080efd85 in dummy_start (data=0x6) at utils.c:538
ASTERISK-5  0x00822ce1 in pthread_start_thread () from /lib/i686/libpthread.so.0
ASTERISK-6 0x001d6b3a in clone () from /lib/i686/libc.so.6


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

jabber.conf

[general]
debug=yes

[asterisk]
type=client
serverhost=talk.google.com
username=asteriskatmisak@gmail.com/asterisk
secret=hidden
port=5222
usetls=yes
usesasl=yes
timeout=100

jingle.conf

[general]
context=gtalk
allowguest=yes

[guest]
disallow=all
allow=alaw
allow=ulaw
context=gtalk

[misaksen]
username=misaksen@gmail.com
disallow=all
allow=alaw
allow=ulaw
context=gtalk
connection=asterisk

Comments:By: Anthony LaMantia (alamantia) 2006-10-16 17:49:22

This looks to be related to 7764
By: Anthony LaMantia (alamantia) 2006-10-16 17:58:02

please try the patch located(from issue 7764) here and report back your findings..

http://bugs.digium.com/file_download.php?file_id=11867&type=bug
By: Morten Isaksen (misaksen) 2006-10-17 14:57:06

Still the same...

   -- Executing [1101@sipclients:1] Dial("SIP/1014-09d1aba0", "Gtalk/asterisk/misaksen@gmail.com") in new stack


#0  0x00c9e069 in gtalk_alloc (client=0x853f258, us=0x84d384a "asteriskatmisak@gmail.com/asterisk3608FABF",
   them=0x7834819 "misaksen@gmail.com", sid=0x0) at chan_gtalk.c:842
842                             resources = client->buddy->resources;
(gdb) bt
#0  0x00c9e069 in gtalk_alloc (client=0x853f258, us=0x84d384a "asteriskatmisak@gmail.com/asterisk3608FABF",
   them=0x7834819 "misaksen@gmail.com", sid=0x0) at chan_gtalk.c:842
#1  0x00c9f9e6 in gtalk_request (type=0x78348e0 "Gtalk", format=8, data=0x7834819, cause=0x7834b18) at chan_gtalk.c:1492
#2  0x0807b4e0 in ast_request (type=0x78348e0 "Gtalk", format=8, data=0x0, cause=0x0) at channel.c:2852
#3  0x0067259e in dial_exec_full (chan=0x85869e8, data=0x78348e0, peerflags=0x7835918) at app_dial.c:1078
#4  0x00676d8e in dial_exec (chan=0x0, data=0x6) at app_dial.c:1653
ASTERISK-1  0x080c1e96 in pbx_extension_helper (c=0x85869e8, con=0x0, context=0x8586b68 "sipclients", exten=0x8586bb8 "1101", priority=1,
   label=0x0, callerid=0x856e8f8 "1014", action=139566620) at pbx.c:505
ASTERISK-2  0x080c31af in __ast_pbx_run (c=0x85869e8) at pbx.c:2245
ASTERISK-3  0x080c509e in pbx_thread (data=0x0) at pbx.c:2556
ASTERISK-4  0x080f0215 in dummy_start (data=0x6) at utils.c:544
ASTERISK-5  0x0035fce1 in pthread_start_thread () from /lib/i686/libpthread.so.0
ASTERISK-6 0x00741b3a in clone () from /lib/i686/libc.so.6
By: Anthony LaMantia (alamantia) 2006-10-18 16:46:35

Can you tell me if you see this error in your error logs?



ast_log(LOG_WARNING, "Out of RTP sessions?\n");
By: Morten Isaksen (misaksen) 2006-10-18 16:54:25

Strange. It is actually working now with audio both ways when I call from Asterisk to gtalk.

The only changes I have made is:

- disabled TLS in jabber.conf
- applied the patch from issue 8164
By: Anthony LaMantia (alamantia) 2006-10-18 16:56:02

also, please try the patch i just uploaded.
By: Anthony LaMantia (alamantia) 2006-10-18 16:59:29

humm sounds, good are you sure it was not the patch from 7764 ? or did you have bindaddr set in your gtalk.conf file ?

also gnuTLS which is used by res_jabbers dependanies have been causing quite some issue with gtalk, removing it seems to solve a heap of problems.

mog has been working converting it to use openssl and working with the team who developed the dependant libray on this issue.
By: Morten Isaksen (misaksen) 2006-10-18 17:21:47

Your patch is applied and it still works. :-)

I just tried to enable TLS and comment out bindaddr and it still worked besides that I got no audio when commented out bindaddr.

I dont know why it did not work before.

Just close this bug.
By: Anthony LaMantia (alamantia) 2006-10-19 09:57:24

resolved with patch from 8164 and turning off tls..