[Asterisk-Dev] *-OH323 - segfault- ast_smoother_feed
Michael Manousos
manousos at inaccessnetworks.com
Tue Jul 15 07:29:23 MST 2003
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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