[Asterisk-Dev] *-OH323 - segfault- ast_smoother_feed

Mon Jul 14 11:58:29 MST 2003

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