[Asterisk-Dev] *-OH323 - segfault- ast_smoother_feed
Hemant Kumar
hemant at versaplanet.com
Mon Jul 14 13:26:00 MST 2003
check the codecs
----- Original Message -----
From: Santosh Prasad <sprasad at hubris.net>
To: <asterisk-dev at lists.digium.com>
Sent: Tuesday, July 15, 2003 12:55 AM
Subject: Re: [Asterisk-Dev] *-OH323 - segfault- ast_smoother_feed
> Hello
>
> > in modules.conf:
> > noload => chan_oss.so
>
> I added noload => chan_oss.so and I I don't get the warning but it
> doesn't help much. I have * crashing when call is placed between
> H323-->SIP endpoints.
>
> Thanks again
>
> Santosh
>
>
> > On Monday 14 July 2003 19:58, Santosh Prasad wrote:
> > > Hello
> > >
> > > I am trying to set up the following scenario:
> > >
> > > SIP(ATA 186)--Asterisk---[OH323-Asterisk-0.5.3]---H323(ATA
> > > 186)---GNUGK(OPENH323GK)
> >
> --------------------------------------------------------------------------
-
> > >-------- I am using the following versions:
> > > Asterisk CVS-07/10/03-12:14:02 built by root at XXX on a i686 running
Linux
> > > Gatekeeper(GNU) Version(2.0.5)
> >
> --------------------------------------------------------------------------
-
> > >-------- When Asterisk loads I get the following warning:
> > >
> > > [chan_oss.so] => (OSS Console Channel Driver)
> > > WARNING[16384]: File chan_oss.c, Line 974 (load_module): XXX I don't
> > > work right with non-full duplex sound cards XXX
> > > == Registered channel type 'Console' (OSS Console Channel Driver)
> > > == Parsing '/etc/asterisk/oss.conf': Found
> > > [New Thread 98311 (LWP 29710)]
> > > [chan_modem_bestdata.so]WARNING[98311]: File chan_oss.c, Line 232
> > > (sound_thread): Read error on sound device: Resource
> > > temporarily unavailable
> >
> --------------------------------------------------------------------------
-
> > >------- I call place calls between SIP endpoints and also between H323
end
> > > points. But when I call from H323 end point to SIP
> > > end point I get a seg fault the gdb is shown below:
> > >
> > > WrapH323EndPoint::AnswerCall: Call with token
> > > ip$207.178.96.112:2118/23558 answered
> > > NOTICE[327701]: File rtp.c, Line 239 (process_rfc3389): RFC3389
support
> > > incomplete. Turn off on client if possible
> > >
> > > Program received signal SIGSEGV, Segmentation fault.
> > > [Switching to Thread 327701 (LWP 29616)]
> > > ast_smoother_feed (s=0xcde9fa29, f=0x81300e8) at frame.c:72
> > > 72 if (!s->format) {
> > > (gdb) bt
> > > #0 ast_smoother_feed (s=0xcde9fa29, f=0x81300e8) at frame.c:72
> > > #1 0x4084b330 in oh323_write (c=0x812d2d0, f=0x81300e8) at
> > > chan_oh323.c:1080
> > > #2 0x080581af in ast_write (chan=0x812d2d0, fr=0x81300e8) at
> > > channel.c:1359
> > > #3 0x0805a541 in ast_channel_bridge (c0=0x81300e8, c1=0x81300e8,
> > > flags=0,
> > > fo=0xbd1feeb4, rc=0xbd1feeb8) at channel.c:2184
> > > #4 0x4022cd3a in ast_bridge_call (chan=0x812d2d0, peer=0x8132a60,
> > > allowredirect_in=0, allowredirect_out=0,
> > > allowdisconnect=0) at res_parking.c:215
> > > #5 0x4068bf4b in dial_exec (chan=0x812d2d0, data=0x4068d05b) at
> > > app_dial.c:648
> > > #6 0x08060d9a in pbx_exec (c=0x812d2d0, app=0x80ea3d0,
data=0xbd1ff74c,
> > > newstack=1) at pbx.c:388
> > > #7 0x08067c38 in pbx_extension_helper (c=0x812d2d0, context=0x80ea3d0
> > > "Dial", exten=0x812d4c0 "5011", priority=1,
> > > callerid=0x8117ab8 "4050", action=135451344) at pbx.c:1130
> > > #8 0x08062bfc in ast_pbx_run (c=0x812d2d0) at pbx.c:1614
> > > #9 0x080682f1 in pbx_thread (data=0x8117dc0) at pbx.c:1830
> > > #10 0x4002f463 in pthread_start_thread () from /lib/libpthread.so.0
> > > #11 0x4002f4df in pthread_start_thread_event () from
> > > /lib/libpthread.so.0
> > > (gdb) print *s
> > > Cannot access memory at address 0xcde9fa29
> > > (gdb) print *f
> > > $1 = {frametype = 2, subclass = 4, datalen = 80, samples = 80, mallocd
=
> > > 0, offset = 76, src = 0x80a9999 "RTP",
> > > data = 0x813015c, prev = 0x0, next = 0x0}
> > > (gdb)
> >
> --------------------------------------------------------------------------
-
> > >-------- when I call from SIP endpoint to H323 endpoint I get the
following
> > > warning and the H323 endpoint doesn't ring:
> > >
> > > Executing Dial("SIP/5010-38a7", "H323/4050 at 207.178.96.112") in new
> > > stack
> > > WARNING[294931]: File channel.c, Line 1546 (ast_request): No channel
> > > type registered for 'H323'
> > > NOTICE[294931]: File app_dial.c, Line 489 (dial_exec): Unable to
create
> > > channel of type 'H323'
> > > == Everyone is busy at this time
> > > -- Timeout on SIP/5010-38a7
> > >
> > > PLAYS DEMO MESSAGE AND EXITS
> > >
> > > NOTICE[294931]: File rtp.c, Line 239 (process_rfc3389): RFC3389
support
> > > incomplete. Turn off on client if possible
> > > -- Executing Hangup("SIP/5010-38a7", "") in new stack
> > > == Spawn extension (voip, #, 2) exited non-zero on 'SIP/5010-38a7'
> > >
> >
> --------------------------------------------------------------------------
-
> > >------------------------------- Selected entries of .conf files are
attached
> > > below
> > >
> > >
> > > extensions.conf is as below:
> > >
> > > [voip]
> > > include => default
> > > exten => s,1,Wait,1 ; Wait a second, just for fun
> > > exten => s,2,Answer ; Answer the line
> > > exten => _XXXX,1,Goto,BYEXTENSION|1
> > > exten => 5010,1,Dial,SIP/5010 at 207.178.96.108
> > > exten => 5011,1,Dial,SIP/5011 at 207.178.96.108
> > > exten => 4050,1,Dial,H323/4050 at 207.178.96.112
> > > exten => 4051,1,Dial,H323/4051 at 207.178.96.112
> > > exten => t,1,Goto(#,1) ; If they take too long, give
up
> > > exten => i,1,Playback(invalid)
> >
> --------------------------------------------------------------------------
-
> > >------------------------
> > >
> > > oh323.conf is as below:
> > >
> > > context=voip
> > > [register]
> > > gwprefix=40
> > > gwprefix=50
> > > context=voip
> > > alias=Asterisk
> > > alias=1010
> > > [codecs]
> > > codec=G711U
> > > frames=20
> >
> --------------------------------------------------------------------------
-
> > >------------------------ sip.conf is as below:
> > >
> > > [5010]
> > > type=friend
> > > username=5010
> > > secret=5050
> > > host=dynamic
> > > defaultip=207.178.96.108
> > >
> > > [5011]
> > > type=friend
> > > username=5011
> > > secret=5051
> > > host=dynamic
> > > defaultip=207.178.96.108
> > >
> >
> --------------------------------------------------------------------------
-
> > >----------------------- Any help would be appreciated.
> > >
> > >
> > >
> > > Thanks
> > >
> > > Santosh
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Dev mailing list
> > > Asterisk-Dev at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> > --
> > Michael Bielicki
> > Managing Director
> > TAAN Consultants Ltd
> > http://www.global-gateway.net/
> >
>
> --------------------------------------------------------------------------
> >
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