[Asterisk-video] asterisk Segmentation fault

[lizhong zhu]

hello,
I am using Sip phone call to my stream server, the dialplan is:
[from-pstn]
exten => s,1,Dial(zap/1)
exten => s,2,Hangup
[from-internal]
exten => 200,1,Dial(SIP/200)
exten => 200,2,Hangup
exten => 2000,1,Dial(SIP/2000)
exten => 2000,2,Hangup
exten => 201,1,Answer
exten => 201,2,Set(SIP_CODEC=h263)
exten => 201,3,Noop(${SIP_CODEC})
exten => 201,4,transcode(,s at camera,h263 at qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50)
exten => 201,4,h324m_loopback()
exten => 201,5,HangUp
[camera]
exten => s,1,Answer
exten => s,2,rtsp(rtsp://192.168.2.160/test2.3gp)
;exten => s,2,rtsp(rtsp://stream.the.sk/live/musicbox/musicbox-hm.3gp)
exten => s,3,HangUp
-------------------------the errors is -----------------------------------
*CLI> [Jan 12 17:58:48] DEBUG[4810]: chan_zap.c:6906 do_monitor: Message status for 6001 changed from -1 to 0 on 1
[Jan 12 17:58:49] DEBUG[4810]: chan_zap.c:6906 do_monitor: Message status for 6002 changed from -1 to 0 on 2
    -- Saved useragent "Kapanga Softphone Desktop 1.00/2174d+1229405425_00138FFD9C26_005056C00001_005056C00008" for peer 200
[Jan 12 17:59:11] NOTICE[4805]: chan_sip.c:15092 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 200
[Jan 12 17:59:11] NOTICE[4805]: chan_sip.c:15092 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 200
    -- Executing [201 at from-internal:1] Answer("SIP/200-08ae9408", "") in new stack
    -- Executing [201 at from-internal:2] Set("SIP/200-08ae9408", "SIP_CODEC=h263") in new stack
    -- Executing [201 at from-internal:3] NoOp("SIP/200-08ae9408", "h263") in new stack
    -- Executing [201 at from-internal:4] transcode("SIP/200-08ae9408", "|s at camera|h263 at qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50") in new stack
    -- Executing [s at camera:1] Answer("Local/s at camera-208d,2", "") in new stack
    -- Executing [s at camera:2] rtsp("Local/s at camera-208d,2", "rtsp://192.168.2.160/test2.3gp") in new stack
[Jan 12 17:59:16] WARNING[4819]: app_rtsp.c:1037 rtsp_play: >rtsp play
[Jan 12 17:59:16] WARNING[4819]: app_rtsp.c:1037 rtsp_play: >rtsp play
[Jan 12 17:59:16] DEBUG[4819]: app_rtsp.c:290 GetUdpPorts: -GetUdpPorts [32778,32779]
[Jan 12 17:59:16] DEBUG[4819]: app_rtsp.c:290 GetUdpPorts: -GetUdpPorts [32780,32781]
[Jan 12 17:59:16] WARNING[4818]: app_transcoder.c:966 app_transcode: >Transcoding [,s at camera,h263 at qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50]
[Jan 12 17:59:16] WARNING[4818]: app_transcoder.c:966 app_transcode: >Transcoding [,s at camera,h263 at qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50]
[Jan 12 17:59:16] DEBUG[4818]: app_transcoder.c:497 VideoTranscoderCreate: -Transcoder [f=0,fps=10,kb=53248,qmin=4,qmax=12,gs=50]
Segmentation fault (core dumped)
----------------------------gdn error-----------------------------------
Loaded symbols for /usr/lib/asterisk/modules/codec_g723-ast14-icc-glibc-pentium4-sse3.so
Reading symbols from /usr/lib/asterisk/modules/app_zapateller.so...done.
Loaded symbols for /usr/lib/asterisk/modules/app_zapateller.so
Reading symbols from /usr/lib/asterisk/modules/app_channelredirect.so...done.
Loaded symbols for /usr/lib/asterisk/modules/app_channelredirect.so
Reading symbols from /usr/lib/asterisk/modules/app_privacy.so...done.
Loaded symbols for /usr/lib/asterisk/modules/app_privacy.so
Reading symbols from /usr/lib/asterisk/modules/app_zapscan.so...done.
Loaded symbols for /usr/lib/asterisk/modules/app_zapscan.so
Reading symbols from /usr/lib/asterisk/modules/app_rtsp.so...done.
Loaded symbols for /usr/lib/asterisk/modules/app_rtsp.so
Reading symbols from /usr/lib/asterisk/modules/app_flash.so...done.
Loaded symbols for /usr/lib/asterisk/modules/app_flash.so
Reading symbols from /usr/lib/asterisk/modules/app_stack.so...done.
Loaded symbols for /usr/lib/asterisk/modules/app_stack.so
Core was generated by `asterisk -vvvvvvvvvvvvvvvgc'.
Program terminated with signal 11, Segmentation fault.
#0  0x01186b11 in av_opt_set_defaults2 () from /usr/lib/libavcodec.so.52
(gdb)
----------------------------------------------------------
anyone has an idea for that problem? it is really troublesome to make a video calls using rtsp. please give me some hints for rtsp.
thanks!
james