[Asterisk-Dev] asterisk-oh323 Bugs (perhaps)

[Rattana BIV]

This is core Backtrace :

(gdb) bt
#0  0x4003d851 in __linuxthreads_create_event () at events.c:26
#1  0x4003a168 in __pthread_create_2_1 (thread=0x80b8d58, attr=0x0,
    start_routine=0x807b0b0 <listener>, arg=0x0) at pthread.c:647
#2  0x0807b521 in ast_makesocket () at asterisk.c:298
#3  0x0807d5a3 in main (argc=1, argv=0xbffffa84) at asterisk.c:1354
#4  0x400d8507 in __libc_start_main (main=0x807cf90 <main>, argc=1,
    ubp_av=0xbffffa84, init=0x80504a4 <_init>, fini=0x809fc30 <_fini>,
    rtld_fini=0x4000dc14 <_dl_fini>, stack_end=0xbffffa7c)
    at ../sysdeps/generic/libc-start.c:129



And on the asterisk CLI I have this :

-- Executing Dial("CAPI[contr1/25]/0", "OH323/192.168.1.22") in new stack
Wrapper API::h323_make_call: Making call.
WrapH323EndPoint::MakeCall: Making call to 192.168.1.22
-- started pbx on channel!
WrapH323EndPoint::CreateConnection: Creating a H323Connection [14512]
WrapH323Connection::WrapH323Connection: WrapH323Connection created.
WrapH323Connection::OnSendSignalSetup: Sending SETUP message...
WrapH323Connection::OnAlerting: Ringing phone for "192.168.1.22" ...
-- Called 192.168.1.22
WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 8, FrameTime 8,
TimeUnits 8
WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160
WrapH323EndPoint::OpenAudioChannel: Frames 20
WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-uLaw-64k
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
PAsteriskSoundChannel::Open: os_handle 37, mediaFormat 0, frameTime 1,
frameNum 20
WrapH323EndPoint::OnStartLogicalChannel: Started logical channel [14512] :
sending G.711-uLaw-64k{hw}
WrapH323EndPoint::OnStartLogicalChannel: TxFrames = 20
WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 8, FrameTime 8,
TimeUnits 8
WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 256
WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-uLaw-64k
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
PAsteriskSoundChannel::Open: os_handle 35, mediaFormat 0, frameTime 1,
frameNum 20
WrapH323EndPoint::OnStartLogicalChannel: Started logical channel [14512] :
receiving G.711-uLaw-64k{hw}
WrapH323EndPoint::OnStartLogicalChannel: RxFrames = 32
-- H323:14512 answered CAPI[contr1/25]/0
-- Setting up echo canceller (PLCI=0x601, function=1, options=2, tail=64)
-- Echo canceller successfully set up (PLCI=0x601)
WrapH323EndPoint::ClearCall: Request to clear call with token
ip$localhost/14512WrapH323Connection::OnReceivedReleaseComplete: Received
RELEASE COMPLETE message...
WrapH323EndPoint::ClearCall: Request to clear call with token
ip$localhost/14512WrapH323EndPoint::ClearCall: Request to clear call with
token ip$localhost/14512WrapH323EndPoint::OnClosedLogicalChannel: Closed
logical channel [14512].
WrapH323EndPoint::OnClosedLogicalChannel: Closed logical channel [14512].
PAsteriskSoundChannel::PAsteriskSoundChannel: Object deleted.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object deleted.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
WrapH323EndPoint::OnConnectionCleared: Closing connection [14512].
WrapH323EndPoint::OnConnectionCleared: "BIV Rattana [192.168.1.22]" has
cleared the call
WrapH323EndPoint::OnConnectionCleared: Duration  0:02
WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.
Segmentation fault


----- Original Message ----- 
From: "Michael Manousos" <manousos at inaccessnetworks.com>
To: <asterisk-dev at lists.digium.com>
Sent: Wednesday, October 08, 2003 12:34 PM
Subject: Re: [Asterisk-Dev] asterisk-oh323 Bugs (perhaps)


>
> Run asterisk with "-vvvcg".
> Do your test (core file generated).
> Run "gdb /usr/sbin/asterisk <core_filename>"
>  From within gdb run "bt" and send me the output
> of it.
>
> Michael.
>
>
>
> Rattana BIV wrote:
> > Where can I find it ?
> >
> > ----- Original Message ----- 
> > From: "Michael Manousos" <manousos at inaccessnetworks.com>
> > To: <asterisk-dev at lists.digium.com>
> > Sent: Wednesday, October 08, 2003 11:59 AM
> > Subject: Re: [Asterisk-Dev] asterisk-oh323 Bugs (perhaps)
> >
> >
> >
> >>This doesn't help.
> >>I 'll need the core backtrace.
> >>
> >>Michael.
> >>
> >>
> >>Rattana BIV wrote:
> >>
> >>>This Segmentation fault happens just when I call Netmeeting with a
> >
> > phone, On
> >
> >>>hangUp i have segmentation fault. (Notice: I use chan_capi for phones)
> >>>
> >>>I put trace in join document...
> >>>
> >>>----- Original Message ----- 
> >>>From: "Michael Manousos" <manousos at inaccessnetworks.com>
> >>>To: <asterisk-dev at lists.digium.com>
> >>>Sent: Wednesday, October 08, 2003 11:21 AM
> >>>Subject: Re: [Asterisk-Dev] asterisk-oh323 Bugs (perhaps)
> >>>
> >>>
> >>>
> >>>
> >>>>Rattana BIV wrote:
> >>>>
> >>>>
> >>>>>Hi,
> >>>>>
> >>>>>
> >>>>>I notice something with the asterisk-oh323 channel drivers.
> >>>>>When I put more than 20 alias asterisk do a Segmentation Fault...
> >>>>>It is normal ? Is there a tip to avoid this ?
> >>>>
> >>>>No, this is not normal. There is no such limitation.
> >>>>Possibly it is something else. Can you provide a core backtrace?
> >>>>
> >>>>
> >>>>
> >>>>>Regards
> >>>>>Rattana
> >>>>
> >>>>Michael.
> >>>>
> >>>>_______________________________________________
> >>>>Asterisk-Dev mailing list
> >>>>Asterisk-Dev at lists.digium.com
> >>>>http://lists.digium.com/mailman/listinfo/asterisk-dev
> >>
> >>
> >>_______________________________________________
> >>Asterisk-Dev mailing list
> >>Asterisk-Dev at lists.digium.com
> >>http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> >
> > _______________________________________________
> > Asterisk-Dev mailing list
> > Asterisk-Dev at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
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