[Asterisk-Dev] *-OH323 - segfault- ast_smoother_feed
Santosh Prasad
sprasad at hubris.net
Tue Jul 15 16:30:52 MST 2003
Hello,
I made the corrections as suggested but does not help much. I still get
the seg fault. Since ast_smoother_feed deals with frames I tried
changing the value of "frames" in oh323.conf file:
[codecs]
codec=G711U
frames=2
;frames=20
I notice that when I place call from SIP end to H323 end I don't get a
seg fault but SIP end doesnot receive any voice packets. But it fails
the otherway. All of the end points used are ATA 186 and configured for
G.711U codec. I would also like to know if the AudioMode value in ATA
186 configuration 0x00140014 in correct for dtmfmode=rfc2833 in sip.conf
and inBandDTMF=no in oh323.conf
Any help is appreciated.
Thanks
Santosh
On Tue, Jul 15, 2003 at 05:29:23PM +0300, Michael Manousos wrote:
>
> Hi,
>
> 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:
>
> I 'm not able to reproduce this error. I can make calls
> to SIP endpoints and receive calls from SIP phones
> without any problem.
>
> >
> > 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
>
> Replace H323 with OH323.
>
> > 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
> >
>
>
>
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