[Asterisk-Dev] *-OH323 - segfault- ast_smoother_feed
Michael Bielicki
Michael.Bielicki at Global-Gateway.net
Mon Jul 14 12:06:05 MST 2003
in modules.conf:
noload => chan_oss.so
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/
--------------------------------------------------------------------------
This correspondence is for the named person's use only. It may contain
confidential or legally privileged information or both. No confidentiality
or privilege is waived or lost by any mistransmission. If you receive this
correspondence in error, please immediately delete it from your system and
notify the sender. You must not disclose, copy or rely on any part of this
correspondence if you are not the intended recipient.
Any opinions expressed in this message are those of the individual sender.
More information about the asterisk-dev
mailing list