asterisk users - Oct 2013 - When i do Video call from sipml5 to sipml5, Call get rejected

Hello All,

I am using Asterisk 12 and sipml5 as front-end and when i call from one 
to another the call will ring on other end but when i allow the camera 
access call will terminated automatically. I have attached the logs of 
Asterisk, if some one will get something useful Please reply on the same.


Thanks and Regards,
Anant



  == Using SIP VIDEO CoS mark 6
   == Using SIP RTP CoS mark 5
[Oct 24 19:45:59] ERROR[3005][C-00000000]: netsock2.c:269 
ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid",
"(null)",
...): Name or service not known
[Oct 24 19:45:59] WARNING[3005][C-00000000]: chan_sip.c:16067 
__set_address_from_contact: Invalid host name in Contact: (can't resolve 
in DNS) : 'df7jal23ls0d.invalid'
[Oct 24 19:45:59] ERROR[3005][C-00000000]: netsock2.c:98 
ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[Oct 24 19:45:59] WARNING[3005][C-00000000]: app_dial.c:2423 
dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - 
Subscriber absent)
     -- Called SIP/1060
     -- SIP/1060-00000001 is ringing
     -- Got SIP response 603 "Failed to get local SDP" back from 
192.168.100.71:42822
     -- SIP/1060-00000001 is busy
   == Everyone is busy/congested at this time (2:1/0/1)
     -- Executing [1060 at default:50006] Goto("SIP/1061-00000000", 
"stdexten-BUSY,1") in new stack
     -- Goto (default,stdexten-BUSY,1)
     -- Executing [stdexten-BUSY at default:1] 
VoiceMail("SIP/1061-00000000", "1060,b") in new stack
[Oct 24 19:46:07] WARNING[3003][C-00000000]: chan_sip.c:24402 
handle_response: Remote host can't match request ACK to call 
'2a8263684cfc957e7da826920c0e59cb at 192.168.100.160:5060'. Giving up.
     -- <SIP/1061-00000000> Playing 'vm-theperson.gsm' (language
'en')
     -- <SIP/1061-00000000> Playing 'digits/1.gsm' (language
'en')
     -- <SIP/1061-00000000> Playing 'digits/0.gsm' (language
'en')
     -- <SIP/1061-00000000> Playing 'digits/6.gsm' (language
'en')
     -- <SIP/1061-00000000> Playing 'digits/0.gsm' (language
'en')
     -- <SIP/1061-00000000> Playing 'vm-isonphone.gsm' (language
'en')
     -- <SIP/1061-00000000> Playing 'vm-intro.gsm' (language
'en')
     -- <SIP/1061-00000000> Playing 'beep.gsm' (language
'en')
     -- Recording the message
     -- x=0, open writing: 
/var/spool/asterisk/voicemail/default/1060/tmp/tJ2W4E format: wav49, 
0x7fb880008408
     -- x=1, open writing: 
/var/spool/asterisk/voicemail/default/1060/tmp/tJ2W4E format: gsm, 
0x7fb88000f618
     -- x=2, open writing: 
/var/spool/asterisk/voicemail/default/1060/tmp/tJ2W4E format: wav, 
0x7fb8800244d8
[Oct 24 19:46:23] WARNING[3005][C-00000000]: app.c:1384 
__ast_play_and_record: No audio available on SIP/1061-00000000??
     -- User hung up
   == Spawn extension (default, stdexten-BUSY, 1) exited non-zero on 
'SIP/1061-00000000'
   == WebSocket connection from '192.168.100.71:42822' closed