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List: asterisk-video
Subject: Re: [Asterisk-video] Asterisk crashes when hanging up a call to
From: Juan_Manuel_Coronado_Zúñiga
Date: 2010-02-11 19:57:15
Message-ID: c8d9da6f1002111157h79cd693asae50094b2aaa6ae () mail ! gmail ! com
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Hi Sergio and list,
Sorry for the late response. Regarding the last question: I was using
app_rtsp rev250 from the svn when I experienced the issue.
Already tested the latest rev255 and it works fine (no more seg-faults) but
only with Asterisk 1.6.2.0. When using 1.6.2.1, it hangs up without showing
any errors on the CLI.
I have another question regarding user and password authentication (since I
still use VLC to do the auth part and then re-stream without auth to
Asterisk). This is how the relevant part in extensions.conf is configured:
exten => 554,2,rtsp(rtsp://user:password@192.168.2.40:554)
Where user and password are both plain text. Is this correct or should it
have another format? The CLI keeps showing:
[100211-093247] DEBUG[29060]: app_rtsp.c:1402 rtsp_play: -Receiving describe
[100211-093247] DEBUG[29060]: app_rtsp.c:1410 rtsp_play: -Describe response
code [401]
[100211-093247] ERROR[29060]: app_rtsp.c:1426 rtsp_play: -No Authenticate
header found
Thank you and best regards,
--
Juan Manuel Coronado Z.
On Thu, Feb 11, 2010 at 7:44 AM, Sergio Garcia Murillo <
sergio.garcia@fontventa.com> wrote:
> Hi all,
>
> I have just commited a new version to the repository in which the problem
> is solved. Aslo I have fixed the RTCP RR that were the cause of the problem.
>
> Best regards
> Sergio
>
> Sergio Garcia Murillo escribió:
>
> Hi Juan Manuel,
>
> Which version are u using of app_rtsp? In latest one the 2088 line is:
>
> ast_module_user_remove(u);
>
> I have checked it and in module.h
>
> #define ast_module_user_remove(user)
> __ast_module_user_remove(ast_module_info->self, user)
>
> So the only way I can think of seg faulting is by ast_module_info beeing
> null. Could you check the value of it in the core?
>
> Best regards
> Sergio
>
>
> Best regards
> Sergio
>
>
> Juan Manuel Coronado Zúñiga escribió:
>
> Hi Sergio,
>
> After some tests, I can confirm that this issue doesn't affect rev240 of
> app_rtsp not on Asterisk 1.6.2.0 nor 1.6.2.0, where it behaves fine and
> calls are finished properly.
>
> Hope this feedback will be of some help.
>
> Regards,
>
>
> --
> Juan Manuel Coronado Z.
>
>
> 2010/2/8 Juan Manuel Coronado Zúñiga <juan.m.coronado@gmail.com>
>
> > Sure, that's a better idea :-)
> >
> > Here's the full backtrace for the Asterisk 1.6.2.0 core dump:
> >
> > (gdb) bt full
> > #0 0xb7397004 in app_rtsp (chan=0x9375590, data=0xb7177f14) at
> > app_rtsp.c:2088
> > u = (struct ast_module_user *)
> > 0x935ae70
> > ip = 0x9378268
> > "172.30.0.25"
> > url = 0xb7177f2b
> > "/test"
> > username =
> > 0x0
> > password =
> > 0x0
> > port =
> > 5553
> > __PRETTY_FUNCTION__ =
> > "app_rtsp"
> > #1 0x08104967 in pbx_exec (c=0x9375590, app=0x9112328, data=0xb7177f14)
> > at pbx.c:1348
> > res = <value optimized
> > out>
> > u = (struct ast_module_user *)
> > 0x9148858
> > saved_c_appl =
> > 0x0
> > saved_c_data =
> > 0x0
> > __PRETTY_FUNCTION__ =
> > "pbx_exec"
> > #2 0x0810f680 in pbx_extension_helper (c=0x9375590, con=0x0,
> > context=0x9375800 "pbx1", exten=0x9375850 "553", priority=2,
> > label=0x0, callerid=0x9374440 "227", action=E_SPAWN, found=0xb717a348,
> > combined_find_spawn=1) at pbx.c:3708
> > e = <value optimized out>
> > app = (struct ast_app *) 0x9112328
> > res = <value optimized out>
> > q = {incstack = {0x0 <repeats 128 times>}, stacklen = 0, status =
> > 5, swo = 0x0, data = 0x0,
> > foundcontext = 0x9375800 "pbx1"}
> > passdata = "rtsp://172.30.0.25:5553/test", '\0' <repeats 8163
> > times>
> > matching_action = 0
> > __PRETTY_FUNCTION__ = "pbx_extension_helper"
> > #3 0x0811192d in __ast_pbx_run (c=0x9375590, args=0x0) at pbx.c:4167
> > dst_exten = '\0' <repeats 12 times>, "1�ѷ", '\0' <repeats 60
> > times>, "\020", '\0' <repeats 19 times>, "\230�7\t", '\0' <repeats 44
> > times>, "T�ѷ", '\0' <repeats 20 times>,
> > "��ѷ\f\000\000\000\230�7\000�\237��`���\000\000\000\0008�\027 \
> > ��\031ҷ`���\f\000\000\000\000\000\000\000`���\200\024:\t\f\000\00 \
> > 0\000\200\033\000\000\210�7\t\235�ٷ\230�7\t\210�7\tpv\a\b\210\021\033\b\f\000\000"
> > pos = 4001536
> > digit = 0
> > invalid = <value optimized out>
> > found = 1
> > ---Type <return> to continue, or q <return> to quit---
> > res = 0
> > error = 0
> > __PRETTY_FUNCTION__ = "__ast_pbx_run"
> > #4 0x08113190 in pbx_thread (data=0x9375590) at pbx.c:4544
> > No locals.
> > #5 0x0815327b in dummy_start (data=0x937e898) at utils.c:968
> > __cancel_buf = {__cancel_jmp_buf = {{__cancel_jmp_buf =
> > {154659208, 0, 4001536, -1223187512, -699156598,
> > -747528460}, __mask_was_saved = 0}}, __pad = {0xb717a480, 0x0,
> > 0x0, 0x0}}
> > not_first_call = <value optimized out>
> > ret = <value optimized out>
> > #6 0xb7c984b5 in start_thread () from /lib/i686/cmov/libpthread.so.0
> > No symbol table info available.
> > #7 0xb7d90a5e in clone () from /lib/i686/cmov/libc.so.6
> > No symbol table info available.
> > (gdb)
> >
> >
> >
> > And for 1.6.2.1:
> >
> > (gdb) bt
> > full
> >
> > #0 0xb72f1004 in app_rtsp (chan=0x86f6608, data=0xb70d1f14) at
> > app_rtsp.c:2088
> > u = (struct ast_module_user *)
> > 0x86f8fb0
> >
> > ip = 0x86fab40
> > "172.30.0.25"
> >
> > url = 0xb70d1f2b
> > "/test"
> >
> > username =
> > 0x0
> >
> > password =
> > 0x0
> >
> > port =
> > 5553
> >
> > __PRETTY_FUNCTION__ =
> > "app_rtsp"
> >
> > #1 0x08104cb7 in pbx_exec (c=0x86f6608, app=0x86f94a0, data=0xb70d1f14)
> > at pbx.c:1348
> > res = <value optimized
> > out>
> >
> > u = (struct ast_module_user *)
> > 0x86cefe0
> >
> > saved_c_appl =
> > 0x0
> >
> > saved_c_data =
> > 0x0
> >
> > __PRETTY_FUNCTION__ =
> > "pbx_exec"
> >
> > #2 0x0810f850 in pbx_extension_helper (c=0x86f6608, con=0x0,
> > context=0x86f6878 "pbx1", exten=0x86f68c8 "553", priority=2,
> > label=0x0, callerid=0x86f91f8 "227", action=E_SPAWN, found=0xb70d4348,
> > combined_find_spawn=1) at pbx.c:3706
> > e = <value optimized
> > out>
> >
> > app = (struct ast_app *)
> > 0x86f94a0
> >
> > res = <value optimized
> > out>
> >
> > q = {incstack = {0x0 <repeats 128 times>}, stacklen = 0, status =
> > 5, swo = 0x0, data = 0x0,
> > foundcontext = 0x86f6878 "pbx1"}
> > passdata = "rtsp://172.30.0.25:5553/test", '\0' <repeats 5808
> > times>, "c^Ç·", '\0' <repeats 12 times>,
> > "x=\r�\000\000\000\000\000=\r�\000\000\000\000\000\000\000\000\a\000\000\000\006",
> > '\0' <repeats 11 times>,
> > "�?\000\000\000\000\2268\213�H<\r��\211ķl<\r�\2268\213�", '\0' \
> > <repeats 20 times>, \
> > "����5\003\000\000\000\000\000\000\000\000\000\000,�ķ", '\0' <repeats \
> > 24 times>, "�;\r�\000\000\000\000\000\000\000\000\000=\r�", '\0' <repeats \
> > 16 times>, "����", '\0' <repeats 40 times>, "����\225"... \
> > matching_action = 0 __PRETTY_FUNCTION__ = "pbx_extension_helper"
> > #3 0x08111b9d in __ast_pbx_run (c=0x86f6608, args=0x0) at pbx.c:4165
> > dst_exten =
> > "\022\000\000\000`ظ�����1\235Ç·\224���t��\001\000\000\000����.ï¿½Ë \
> > �\201\\C\r��\201��LC\r�t��@C\r�����\000\000\000\000`Ø¸ï¿ \
> > ½\005\000\000\000\020\000\000\000\001\000\000\000pC\r��;�dC\r��Uo\b\000 \
> > \000\000\000@ڸ�LC\r�@C\r�\001\000\000\000<C\r�\000\000\000\000\214C\r�\000\200\a\237�\000\000\000\000T\217Ƿ",
> > '\0' <repeats 12 times>,
> > "@C\r��C\r��{Ƿ\f\000\000\000�Uo\000�?`Q\000\000\000\0008C\r���Ƿ`Q"...
> >
> > ---Type <return> to continue, or q <return> to quit---
> > pos = 4001536
> > digit = 0
> > invalid = <value optimized out>
> > found = 1
> > res = 0
> > error = 0
> > __PRETTY_FUNCTION__ = "__ast_pbx_run"
> > #4 0x08113400 in pbx_thread (data=0x86f6608) at pbx.c:4542
> > No locals.
> > #5 0x0815305b in dummy_start (data=0x86f55a0) at utils.c:968
> > __cancel_buf = {__cancel_jmp_buf = {{__cancel_jmp_buf =
> > {141435248, 0, 4001536, -1223867448, 161732390, 960872024},
> > __mask_was_saved = 0}}, __pad = {0xb70d4480, 0x0, 0xb70d43c4,
> > 0xb7d63ff4}}
> > not_first_call = <value optimized out>
> > ret = <value optimized out>
> > #6 0xb7bf24b5 in start_thread () from /lib/i686/cmov/libpthread.so.0
> > No symbol table info available.
> > #7 0xb7ceaa5e in clone () from /lib/i686/cmov/libc.so.6
> > No symbol table info available.
> > (gdb)
> >
> >
> > Best regards,
> >
> > --
> > Juan Manuel Coronado Z.
> >
> >
> >
> > On Mon, Feb 8, 2010 at 11:39 AM, Sergio Garcia Murillo <
> > sergio.garcia@fontventa.com> wrote:
> >
> > > Hi Juan Manuel,
> > >
> > > Could you send a backtrace of the core dumps instead?
> > >
> > > Best regards
> > > Sergio
> > >
> > > Juan Manuel Coronado Zúñiga escribió:
> > > >
> > > > Hi Sergio and list,
> > > >
> > > > Hanging up a succesfull call from a Eyebeam Softphone (H.264) to
> > > > app_rtsp causes Asterisk to crash. I've tried versions 1.6.0.10,
> > > > 1.6.2.0 and 1.6.2.1.
> > > >
> > > > I have the following setup:
> > > >
> > > > - Asterisk with app_rtsp rev 250 in Debian Lenny (5.0.3) i386.
> > > > - Grandstream GXV3601 registered in Asterisk.
> > > > - VLC in other PC which connects to the Camera, performs the
> > > > authentication (I've experienced issues with app_rtsp auth) and
> > > > re-streams to Asterisk.
> > > > - Windows machine with Eyebeam softphone and a webcam attached.
> > > >
> > > > The call to app_rtsp succesfully shows the video streamed by the
> > > > camera, but when hanging up it causes a segfault. I can provide 2 core
> > > > dumps with the latest versions of Asterisk if needed (1.3 MB).
> > > >
> > > > Any help would be appreciated.
> > > >
> > > > Regards,
> > > >
> > > >
> > > > --
> > > > Juan Manuel Coronado Z.
> > > >
> > >
> > >
> > >
> > > --
> > > _____________________________________________________________________
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > >
> > > asterisk-video mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-video
> >
> >
> >
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
[Attachment #5 (text/html)]
Hi Sergio and list,<br><br>Sorry for the late response. Regarding the last question: \
I was using app_rtsp rev250 from the svn when I experienced the issue.<br><br>Already \
tested the latest rev255 and it works fine (no more seg-faults) but only with \
Asterisk 1.6.2.0. When using 1.6.2.1, it hangs up without showing any errors on the \
CLI.<br>
<br>I have another question regarding user and password authentication (since I still \
use VLC to do the auth part and then re-stream without auth to Asterisk). This is how \
the relevant part in extensions.conf is configured:<br>
<br>exten => 554,2,rtsp(rtsp://<a \
href="http://user:password@192.168.2.40:554">user:password@192.168.2.40:554</a>)<br><br>Where \
user and password are both plain text. Is this correct or should it have another \
format? The CLI keeps showing:<br>
<br>[100211-093247] DEBUG[29060]: app_rtsp.c:1402 rtsp_play: -Receiving \
describe<br>[100211-093247] DEBUG[29060]: app_rtsp.c:1410 rtsp_play: -Describe \
response code [401]<br>[100211-093247] ERROR[29060]: app_rtsp.c:1426 rtsp_play: -No \
Authenticate header found<br>
<br>Thank you and best regards,<br><br><br clear="all">--<br>Juan Manuel Coronado \
Z.<br> <br><br><div class="gmail_quote">On Thu, Feb 11, 2010 at 7:44 AM, Sergio \
Garcia Murillo <span dir="ltr"><<a \
href="mailto:sergio.garcia@fontventa.com">sergio.garcia@fontventa.com</a>></span> \
wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, \
204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000">
Hi all,<br>
<br>
I have just commited a new version to the repository in which the
problem is solved. Aslo I have fixed the RTCP RR that were the cause of
the problem.<br>
<br>
Best regards<br>
Sergio<br><font color="#888888">
<br>
Sergio Garcia Murillo escribió:
</font><div><div></div><div class="h5"><blockquote type="cite">
Hi Juan Manuel,<br>
<br>
Which version are u using of app_rtsp? In latest one the 2088 line is:<br>
<br>
ast_module_user_remove(u);<br>
<br>
I have checked it and in module.h<br>
<br>
#define ast_module_user_remove(user)
__ast_module_user_remove(ast_module_info->self, user)<br>
<br>
So the only way I can think of seg faulting is by ast_module_info
beeing null. Could you check the value of it in the core?<br>
<br>
Best regards<br>
Sergio<br>
<br>
<br>
Best regards<br>
Sergio<br>
<br>
<br>
Juan Manuel Coronado Zúñiga escribió:
<blockquote type="cite">Hi Sergio,<br>
<br>
After some tests, I can confirm that this issue doesn't affect rev240
of app_rtsp not on Asterisk 1.6.2.0 nor 1.6.2.0, where it behaves fine
and calls are finished properly.<br>
<br>
Hope this feedback will be of some help.<br>
<br>
Regards,<br>
<br>
<br clear="all">
--<br>
Juan Manuel Coronado Z.<br>
<br>
<br>
<div class="gmail_quote">2010/2/8 Juan Manuel Coronado Zúñiga <span \
dir="ltr"><<a href="mailto:juan.m.coronado@gmail.com" \
target="_blank">juan.m.coronado@gmail.com</a>></span><br> <blockquote \
class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt \
0pt 0.8ex; padding-left: 1ex;">Sure, that's a better idea :-)<br>
<br>
Here's the full backtrace for the Asterisk 1.6.2.0 core dump:<br>
<br>
(gdb) bt full<br>
#0 0xb7397004 in app_rtsp (chan=0x9375590, data=0xb7177f14) at
app_rtsp.c:2088<br>
u = (struct ast_module_user *)
0x935ae70 <br>
ip = 0x9378268
"172.30.0.25" <br>
url = 0xb7177f2b
"/test" <br>
username =
0x0 <br>
password =
0x0 <br>
port =
5553 <br>
__PRETTY_FUNCTION__ =
"app_rtsp" <br>
#1 0x08104967 in pbx_exec (c=0x9375590, app=0x9112328,
data=0xb7177f14) at pbx.c:1348<br>
res = <value optimized
out> <br>
u = (struct ast_module_user *)
0x9148858 <br>
saved_c_appl =
0x0 <br>
saved_c_data =
0x0 <br>
__PRETTY_FUNCTION__ =
"pbx_exec" <br>
#2 0x0810f680 in pbx_extension_helper (c=0x9375590, con=0x0,
context=0x9375800 "pbx1", exten=0x9375850 "553", priority=2, <br>
label=0x0, callerid=0x9374440 "227", action=E_SPAWN,
found=0xb717a348, combined_find_spawn=1) at pbx.c:3708<br>
e = <value optimized out><br>
app = (struct ast_app *) 0x9112328<br>
res = <value optimized out><br>
q = {incstack = {0x0 <repeats 128 times>}, stacklen = 0,
status = 5, swo = 0x0, data = 0x0,<br>
foundcontext = 0x9375800 "pbx1"}<br>
passdata = "rtsp://<a href="http://172.30.0.25:5553/test" \
target="_blank">172.30.0.25:5553/test</a>", '\0' <repeats 8163 \
times><br> matching_action = 0<br>
__PRETTY_FUNCTION__ = "pbx_extension_helper"<br>
#3 0x0811192d in __ast_pbx_run (c=0x9375590, args=0x0) at pbx.c:4167<br>
dst_exten = '\0' <repeats 12 times>, "1�ѷ", \
'\0' <repeats 60 times>, "\020", '\0' <repeats 19 \
times>, "\230�7\t", '\0' <repeats 44 times>, \
"T�ѷ", '\0' <repeats 20 times>,
"��ѷ\f\000\000\000\230�7\000�\237��`���\000\000\000\0008�\027��\031ҷ`���\f\000\00 \
0\000\000\000\000\000`���\200\024:\t\f\000\000\000\200\033\000\000\210�7\t\235�ٷ\230�7\t\210�7\tpv\a\b\210\021\033\b\f\000\000"<br>
pos = 4001536<br>
digit = 0<br>
invalid = <value optimized out><br>
found = 1<br>
---Type <return> to continue, or q <return> to quit---<br>
res = 0<br>
error = 0<br>
__PRETTY_FUNCTION__ = "__ast_pbx_run"<br>
#4 0x08113190 in pbx_thread (data=0x9375590) at pbx.c:4544<br>
No locals.<br>
#5 0x0815327b in dummy_start (data=0x937e898) at utils.c:968<br>
__cancel_buf = {__cancel_jmp_buf = {{__cancel_jmp_buf =
{154659208, 0, 4001536, -1223187512, -699156598,<br>
-747528460}, __mask_was_saved = 0}}, __pad = {0xb717a480, 0x0,
0x0, 0x0}}<br>
not_first_call = <value optimized out><br>
ret = <value optimized out><br>
#6 0xb7c984b5 in start_thread () from /lib/i686/cmov/libpthread.so.0<br>
No symbol table info available.<br>
#7 0xb7d90a5e in clone () from /lib/i686/cmov/libc.so.6<br>
No symbol table info available.<br>
(gdb)<br>
<br>
<br>
<br>
And for <a href="http://1.6.2.1" target="_blank">1.6.2.1</a>:<br>
<br>
(gdb) bt
full \
<br>
#0 0xb72f1004 in app_rtsp (chan=0x86f6608, data=0xb70d1f14) at
app_rtsp.c:2088 <br>
u = (struct ast_module_user *)
0x86f8fb0
<br>
ip = 0x86fab40
"172.30.0.25" \
<br>
url = 0xb70d1f2b
"/test" \
<br>
username =
0x0 \
<br>
password =
0x0 \
<br>
port =
5553 \
<br>
__PRETTY_FUNCTION__ =
"app_rtsp" \
<br>
#1 0x08104cb7 in pbx_exec (c=0x86f6608, app=0x86f94a0,
data=0xb70d1f14) at pbx.c:1348 <br>
res = <value optimized
out> \
<br>
u = (struct ast_module_user *)
0x86cefe0
<br>
saved_c_appl =
0x0 \
<br>
saved_c_data =
0x0 \
<br>
__PRETTY_FUNCTION__ =
"pbx_exec" \
<br>
#2 0x0810f850 in pbx_extension_helper (c=0x86f6608, con=0x0,
context=0x86f6878 "pbx1", exten=0x86f68c8 "553", priority=2, \
<br> label=0x0, callerid=0x86f91f8 "227", action=E_SPAWN,
found=0xb70d4348, combined_find_spawn=1) at pbx.c:3706 <br>
e = <value optimized
out> \
<br>
app = (struct ast_app *)
0x86f94a0 \
<br>
res = <value optimized
out> \
<br>
q = {incstack = {0x0 <repeats 128 times>}, stacklen = 0,
status = 5, swo = 0x0, data = 0x0,<br>
foundcontext = 0x86f6878 "pbx1"}<br>
passdata = "rtsp://<a href="http://172.30.0.25:5553/test" \
target="_blank">172.30.0.25:5553/test</a>", '\0' <repeats 5808 \
times>, "c^Ƿ", '\0' <repeats 12 times>, \
"x=\r�\000\000\000\000\000=\r�\000\000\000\000\000\000\000\000\a\000\000\000\006",
'\0' <repeats 11 times>,
"�?\000\000\000\000\2268\213�H<\r��\211ķl<\r�\2268\213�", \
'\0' <repeats 20 times>,
"����5\003\000\000\000\000\000\000\000\000\000\000,�ķ", '\0'
<repeats 24 times>,
"�;\r�\000\000\000\000\000\000\000\000\000=\r�", '\0' <repeats \
16 times>, "����", '\0' <repeats 40 times>, \
"����\225"...<br> matching_action = 0<br>
__PRETTY_FUNCTION__ = "pbx_extension_helper"<br>
#3 0x08111b9d in __ast_pbx_run (c=0x86f6608, args=0x0) at pbx.c:4165<br>
dst_exten =
"\022\000\000\000`ظ�����1\235Ƿ\224���t��\001\000\000\000����.�ˠ�\201\\C\r��\201�� \
LC\r�t��@C\r�����\000\000\000\000`ظ�\005\000\000\000\020\000\000\000\001\000\000\000pC \
\r��;�dC\r��Uo\b\000\000\000\000@ڸ�LC\r�@C\r�\001\000\000\000<C\r�\000\000\000\000\214C\r�\000\200\a\237�\000\000\000\000T\217Ƿ",
'\0' <repeats 12 times>,
"@C\r��C\r��{Ƿ\f\000\000\000�Uo\000�?`Q\000\000\000\0008C\r���Ƿ`Q"...<br>
---Type <return> to continue, or q <return> to quit---<br>
pos = 4001536<br>
digit = 0<br>
invalid = <value optimized out><br>
found = 1<br>
res = 0<br>
error = 0<br>
__PRETTY_FUNCTION__ = "__ast_pbx_run"<br>
#4 0x08113400 in pbx_thread (data=0x86f6608) at pbx.c:4542<br>
No locals.<br>
#5 0x0815305b in dummy_start (data=0x86f55a0) at utils.c:968<br>
__cancel_buf = {__cancel_jmp_buf = {{__cancel_jmp_buf =
{141435248, 0, 4001536, -1223867448, 161732390, 960872024},<br>
__mask_was_saved = 0}}, __pad = {0xb70d4480, 0x0, 0xb70d43c4,
0xb7d63ff4}}<br>
not_first_call = <value optimized out><br>
ret = <value optimized out><br>
#6 0xb7bf24b5 in start_thread () from /lib/i686/cmov/libpthread.so.0<br>
No symbol table info available.<br>
#7 0xb7ceaa5e in clone () from /lib/i686/cmov/libc.so.6<br>
No symbol table info available.<br>
(gdb)<br>
<br>
<br>
Best regards,<br>
<font color="#888888"><br clear="all">
--<br>
Juan Manuel Coronado Z.</font>
<div>
<div><br>
<br>
<br>
<div class="gmail_quote">On Mon, Feb 8, 2010 at 11:39 AM, Sergio
Garcia Murillo <span dir="ltr"><<a href="mailto:sergio.garcia@fontventa.com" \
target="_blank">sergio.garcia@fontventa.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, \
204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi Juan Manuel,<br>
<br>
Could you send a backtrace of the core dumps instead?<br>
<br>
Best regards<br>
Sergio<br>
<br>
Juan Manuel Coronado Zúñiga escribió:<br>
<div>
<div>><br>
> Hi Sergio and list,<br>
><br>
> Hanging up a succesfull call from a Eyebeam Softphone (H.264) to<br>
> app_rtsp causes Asterisk to crash. I've tried versions 1.6.0.10,<br>
> 1.6.2.0 and 1.6.2.1.<br>
><br>
> I have the following setup:<br>
><br>
> - Asterisk with app_rtsp rev 250 in Debian Lenny (5.0.3) i386.<br>
> - Grandstream GXV3601 registered in Asterisk.<br>
> - VLC in other PC which connects to the Camera, performs the<br>
> authentication (I've experienced issues with app_rtsp auth) and<br>
> re-streams to Asterisk.<br>
> - Windows machine with Eyebeam softphone and a webcam attached.<br>
><br>
> The call to app_rtsp succesfully shows the video streamed by the<br>
> camera, but when hanging up it causes a segfault. I can provide 2
core<br>
> dumps with the latest versions of Asterisk if needed (1.3 MB).<br>
><br>
> Any help would be appreciated.<br>
><br>
> Regards,<br>
><br>
><br>
> --<br>
> Juan Manuel Coronado Z.<br>
><br>
<br>
<br>
<br>
</div>
</div>
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</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
</blockquote>
<br>
</blockquote>
<br>
</div></div></div>
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