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@default:50006] Goto("SIP/1061-00000000",
"stdexten-BUSY,1") in new stack
-- Goto (default,stdexten-BUSY,1)
-- Executing [stdexten-BUSY@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@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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
