[Asterisk-video] Asterisk video and AMI Originate
Joost Kuif | Mobillion
Joost.kuif at mobillion.nl
Tue Feb 19 04:48:07 CST 2008
Hi Sergio,
the backtrace is as follows:
Core was generated by `/usr/sbin/asterisk -f'.
Program terminated with signal 11, Segmentation fault.
#0 0x00002aaac7574c1a in ?? () from
/usr/lib/asterisk/modules/codec_amr.so
(gdb) bt
#0 0x00002aaac7574c1a in ?? () from
/usr/lib/asterisk/modules/codec_amr.so
#1 0x00002aaac757a114 in ?? () from
/usr/lib/asterisk/modules/codec_amr.so
#2 0x00002aaac757c160 in Speech_Decode_Frame (st=0x88c630,
mode=1082058752,
parm=0x1f3e, frame_type=3344492448, synth=0x886f68) at sp_dec.c:5676
#3 0x00002aaac7558e95 in Decoder_Interface_Decode (st=0x88c610,
bits=<value optimized out>, synth=0x886f68, bfi=<value optimized
out>)
at interf_dec.c:816
#4 0x00002aaac7557bc3 in amrtolin_framein (pvt=0x882960, f=<value
optimized out>)
at codec_amr.c:266
#5 0x00000000004a4c21 in framein (pvt=0x882960, f=0x407f0060) at
translate.c:189
#6 0x00000000004a4e4c in ast_translate (path=0x882960, f=0x407f0060,
consume=0)
at translate.c:334
#7 0x0000000000440161 in ast_write (chan=0x85db90, fr=0x407f0060)
at channel.c:2879
#8 0x00002aaabeef0a65 in mp4_rtp_read (p=0x407f07d0) at app_mp4.c:312
#9 0x00002aaabeef0f68 in mp4_play (chan=0x85db90, data=0x65) at
app_mp4.c:474
#10 0x000000000047f960 in pbx_extension_helper (c=0x85db90,
con=<value optimized out>, context=0x85dde0 "xcon", exten=0x85de30
"666",
priority=6, label=<value optimized out>, callerid=0x831c00 "",
action=E_SPAWN)
at pbx.c:532
#11 0x000000000048135b in __ast_pbx_run (c=0x85db90) at pbx.c:2288
#12 0x0000000000481f89 in pbx_thread (data=0xffffffffffffffdb) at
pbx.c:2603
#13 0x00000000004a999c in dummy_start (data=<value optimized out>) at
utils.c:775
---Type <return> to continue, or q <return> to quit---
#14 0x00002b9663d652a5 in start_thread () from /lib/libpthread.so.0
#15 0x00002b9664c5061d in clone () from /lib/libc.so.6
#16 0x0000000000000000 in ?? ()
(gdb)
In my dialplan i have:
[xcon]
exten => 665,1,h324m_call(666 at xcon)
exten => 666,1,Set(CHANNEL(transfercapability)=VIDEO)
exten => 666,n,NoOp(transfer=${CHANNEL(transfercapability)})
exten => 666,n,Set(CHANNEL(userinformationlayer1)=38)
exten => 666,n,NoOp(ul1=${CHANNEL(userinformationlayer1)})
exten => 666,n,Dial(Zap/g1/0123456789)
exten => 666,n,mp4play(/var/lib/asterisk/videos/sample.3gp)
exten => 666,n,Hangup
I have a script which i pipe to telnet
#!/bin/sh
echo "open localhost 5038"
sleep 1
echo "Action: login"
echo "UserName: manager"
echo "Secret: mysecret"
echo
echo "Action: Originate"
#echo "Channel: Zap/g1/0123456789"
echo "Channel: Local/665 at xcon"
#echo "Variable:
CHANNEL(transfercapability)=VIDEO|CHANNEL(userinformationlayer1)=38"
#echo "Application: h324m_call"
echo "Application: h324m_gw"
echo "Data: 665 at xcon"
echo
sleep 10
logging from asterisk is:
root at umts01:/tmp# [Feb 19 11:42:18] DEBUG[6759]: manager.c:2032
process_message: Manager received command 'login'
[Feb 19 11:42:18] DEBUG[6759]: config.c:846 config_text_file_load:
Parsing /etc/asterisk/manager.conf
[Feb 19 11:42:18] DEBUG[6759]: acl.c:200 ast_append_ha:
0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer
[Feb 19 11:42:18] DEBUG[6759]: acl.c:200 ast_append_ha:
127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer
[Feb 19 11:42:18] DEBUG[6759]: acl.c:200 ast_append_ha:
192.168.218.65/255.255.255.0/255.255.255.0 appended to acl for peer
[Feb 19 11:42:18] DEBUG[6759]: acl.c:215 ast_apply_ha: ##### Testing
127.0.0.1 with 0.0.0.0
[Feb 19 11:42:18] DEBUG[6759]: acl.c:215 ast_apply_ha: ##### Testing
127.0.0.1 with 127.0.0.0
[Feb 19 11:42:18] DEBUG[6759]: acl.c:215 ast_apply_ha: ##### Testing
127.0.0.1 with 192.168.218.0
[Feb 19 11:42:18] DEBUG[6759]: manager.c:2032 process_message: Manager
received command 'Originate'
[Feb 19 11:42:18] DEBUG[6759]: chan_zap.c:7842 zt_request: Using channel
1
[Feb 19 11:42:18] DEBUG[6759]: devicestate.c:304
__ast_device_state_changed_literal: Notification of state change to be
queued on device/channel Zap/1-1
[Feb 19 11:42:18] DEBUG[6715]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for Zap - 1
[Feb 19 11:42:18] DEBUG[6715]: devicestate.c:287 do_state_change:
Changing state for Zap/1 - state 2 (In use)
[Feb 19 11:42:18] DEBUG[6761]: app_queue.c:548 changethread: Device
'Zap/1' changed to state '2' (In use) but we don't care because they're
not a member of any queue.
[Feb 19 11:42:18] DEBUG[6740]: chan_zap.c:8982 pri_dchannel: Queuing
frame from PRI_EVENT_PROCEEDING on channel 0/1 span 1
[Feb 19 11:42:22] DEBUG[6740]: chan_zap.c:1417 zt_enable_ec: Echo
cancellation isn't required on digital connection
[Feb 19 11:42:22] DEBUG[6759]: devicestate.c:304
__ast_device_state_changed_literal: Notification of state change to be
queued on device/channel Zap/1-1
[Feb 19 11:42:22] DEBUG[6715]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for Zap - 1
[Feb 19 11:42:22] DEBUG[6715]: devicestate.c:287 do_state_change:
Changing state for Zap/1 - state 6 (Ringing)
[Feb 19 11:42:22] DEBUG[6763]: app_queue.c:548 changethread: Device
'Zap/1' changed to state '6' (Ringing) but we don't care because they're
not a member of any queue.
[Feb 19 11:42:25] DEBUG[6740]: chan_zap.c:1417 zt_enable_ec: Echo
cancellation isn't required on digital connection
[Feb 19 11:42:25] DEBUG[6759]: devicestate.c:304
__ast_device_state_changed_literal: Notification of state change to be
queued on device/channel Zap/1-1
[Feb 19 11:42:25] DEBUG[6715]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for Zap - 1
[Feb 19 11:42:25] DEBUG[6715]: devicestate.c:287 do_state_change:
Changing state for Zap/1 - state 2 (In use)
[Feb 19 11:42:25] WARNING[6759]: pbx.c:5138 ast_pbx_outgoing_app:
Zap/1-1 already has a call record??
[Feb 19 11:42:25] DEBUG[6765]: app_queue.c:548 changethread: Device
'Zap/1' changed to state '2' (In use) but we don't care because they're
not a member of any queue.
[Feb 19 11:42:25] DEBUG[6766]: app_h324m.c:685 app_h324m_gw: h324m_gw
[Feb 19 11:42:25] DEBUG[6767]: pbx.c:1809 pbx_extension_helper:
Launching 'h324m_call'
[Feb 19 11:42:25] DEBUG[6767]: app_h324m.c:992 app_h324m_call:
h324m_call
[Feb 19 11:42:25] DEBUG[6768]: pbx.c:1809 pbx_extension_helper:
Launching 'Set'
[Feb 19 11:42:25] DEBUG[6768]: pbx.c:1662
pbx_substitute_variables_helper_full: Function result is 'VIDEO'
[Feb 19 11:42:25] DEBUG[6768]: pbx.c:1809 pbx_extension_helper:
Launching 'NoOp'
[Feb 19 11:42:25] DEBUG[6768]: pbx.c:1809 pbx_extension_helper:
Launching 'Set'
[Feb 19 11:42:25] DEBUG[6768]: pbx.c:1662
pbx_substitute_variables_helper_full: Function result is '38'
[Feb 19 11:42:25] DEBUG[6768]: pbx.c:1809 pbx_extension_helper:
Launching 'NoOp'
[Feb 19 11:42:25] DEBUG[6768]: pbx.c:1809 pbx_extension_helper:
Launching 'Dial'
[Feb 19 11:42:25] DEBUG[6768]: chan_zap.c:7842 zt_request: Using channel
2
[Feb 19 11:42:25] DEBUG[6768]: rtp.c:1576 ast_rtp_make_compatible:
Channel 'Zap/2-1' has no RTP, not doing anything
[Feb 19 11:42:25] DEBUG[6768]: channel.c:3493
ast_channel_inherit_variables: Not copying variable STACK-xcon-666-5.
[Feb 19 11:42:25] DEBUG[6768]: channel.c:3493
ast_channel_inherit_variables: Not copying variable STACK-xcon-666-4.
[Feb 19 11:42:25] DEBUG[6768]: channel.c:3493
ast_channel_inherit_variables: Not copying variable STACK-xcon-666-3.
[Feb 19 11:42:25] DEBUG[6768]: channel.c:3493
ast_channel_inherit_variables: Not copying variable STACK-xcon-666-2.
[Feb 19 11:42:25] DEBUG[6768]: channel.c:3493
ast_channel_inherit_variables: Not copying variable STACK-xcon-666-1.
[Feb 19 11:42:25] DEBUG[6768]: devicestate.c:304
__ast_device_state_changed_literal: Notification of state change to be
queued on device/channel Zap/2-1
[Feb 19 11:42:25] DEBUG[6715]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for Zap - 2
[Feb 19 11:42:25] DEBUG[6715]: devicestate.c:287 do_state_change:
Changing state for Zap/2 - state 2 (In use)
[Feb 19 11:42:25] DEBUG[6769]: app_queue.c:548 changethread: Device
'Zap/2' changed to state '2' (In use) but we don't care because they're
not a member of any queue.
[Feb 19 11:42:25] DEBUG[6740]: chan_zap.c:8982 pri_dchannel: Queuing
frame from PRI_EVENT_PROCEEDING on channel 0/2 span 1
[Feb 19 11:42:25] DEBUG[6768]: rtp.c:1493 ast_rtp_early_bridge: Channel
'Local/666 at xcon-3d43,2' has no RTP, not doing anything
[Feb 19 11:42:27] DEBUG[6768]: channel.c:1766 ast_hangup: Hanging up
channel 'Zap/2-1'
[Feb 19 11:42:27] DEBUG[6768]: chan_zap.c:2420 zt_hangup:
zt_hangup(Zap/2-1)
[Feb 19 11:42:27] DEBUG[6768]: chan_zap.c:2969 zt_setoption: Set option
AUDIO MODE, value: ON(1) on Zap/2-1
[Feb 19 11:42:27] DEBUG[6768]: chan_zap.c:2454 zt_hangup: Hangup:
channel: 2 index = 0, normal = 26, callwait = -1, thirdcall = -1
[Feb 19 11:42:27] DEBUG[6768]: chan_zap.c:2608 zt_hangup: Not yet
hungup... Calling hangup once with icause, and clearing call
[Feb 19 11:42:27] DEBUG[6768]: chan_zap.c:1466 zt_disable_ec: disabled
echo cancellation on channel 2
[Feb 19 11:42:27] DEBUG[6768]: chan_zap.c:2884 zt_setoption: Set option
TDD MODE, value: OFF(0) on Zap/2-1
[Feb 19 11:42:27] DEBUG[6768]: chan_zap.c:1402 update_conf: Updated
conferencing on 2, with 0 conference users
[Feb 19 11:42:27] DEBUG[6768]: chan_zap.c:2965 zt_setoption: Set option
AUDIO MODE, value: OFF(0) on Zap/2-1
[Feb 19 11:42:27] DEBUG[6768]: chan_zap.c:1466 zt_disable_ec: disabled
echo cancellation on channel 2
[Feb 19 11:42:27] DEBUG[6768]: devicestate.c:304
__ast_device_state_changed_literal: Notification of state change to be
queued on device/channel Zap/2-1
[Feb 19 11:42:27] DEBUG[6715]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for Zap - 2
[Feb 19 11:42:27] DEBUG[6715]: devicestate.c:287 do_state_change:
Changing state for Zap/2 - state 0 (Unknown)
[Feb 19 11:42:27] DEBUG[6770]: app_queue.c:548 changethread: Device
'Zap/2' changed to state '0' (Unknown) but we don't care because they're
not a member of any queue.
[Feb 19 11:42:27] DEBUG[6768]: rtp.c:1493 ast_rtp_early_bridge: Channel
'Local/666 at xcon-3d43,2' has no RTP, not doing anything
[Feb 19 11:42:27] DEBUG[6768]: app_dial.c:1687 dial_exec_full: Exiting
with DIALSTATUS=BUSY.
[Feb 19 11:42:27] DEBUG[6768]: pbx.c:1809 pbx_extension_helper:
Launching 'mp4play'
[Feb 19 11:42:27] DEBUG[6768]: app_mp4.c:360 mp4_play: mp4play
/var/lib/asterisk/videos/video2web/hoofdmenu_1.3gp
mp4play /var/lib/asterisk/videos/video2web/hoofdmenu_1.3gp
ReadAtom: invalid atom size, extends outside parent atom - skipping to
end of "rtng" "MPAA" 56673 vs 210
[Feb 19 11:42:27] DEBUG[6768]: app_mp4.c:389 mp4_play: found hint track
65335
found hint track 65335
[Feb 19 11:42:27] DEBUG[6768]: app_mp4.c:401 mp4_play: track 201 vide
track 201 vide
[Feb 19 11:42:27] DEBUG[6768]: app_mp4.c:389 mp4_play: found hint track
65435
found hint track 65435
[Feb 19 11:42:27] DEBUG[6768]: app_mp4.c:401 mp4_play: track 101 soun
track 101 soun
[Feb 19 11:42:27] DEBUG[6768]: channel.c:3022 set_format: Set channel
Local/666 at xcon-3d43,2 to write format amr
MP4ERROR: FindTrackId: Track index doesn't exist - track 2 type hint
[Feb 19 11:42:27] WARNING[6768]: utils.c:961 tvfix: warning negative
timestamp 0.-4846791580151137040
Segmentation fault (core dumped)
-----Oorspronkelijk bericht-----
Van: asterisk-video-bounces at lists.digium.com
[mailto:asterisk-video-bounces at lists.digium.com] Namens Sergio Garcia
Murillo
Verzonden: Monday, February 18, 2008 7:44 PM
Aan: Development discussion of video media support in Asterisk
Onderwerp: Re: [Asterisk-video] Asterisk video and AMI Originate
Could you post a backtrace of the dump?
I've got it working correctly on my tests.
BR
Sergio
On Mon, 2008-02-18 at 16:49 +0100, Joost Kuif | Mobillion wrote:
> Hi,
>
> I would like to start a outgoing video call. I have tried various ways
> via the AMI interface, which signal an incoming videocall on the
> phone, but all ending up in core dumps in asterisk.
> Has someone already succeeded to video-dial out to a mobile phone from
> asterisk?
>
> Regards,
> Joost
>
> _______________________________________________
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