[Asterisk-video] asterisk-video Digest, Vol 84, Issue 1
bipin singh
bipinraghuvanshi at gmail.com
Thu Oct 24 22:24:19 CDT 2013
On Thu, Oct 24, 2013 at 10:30 PM,
<asterisk-video-request at lists.digium.com>wrote:
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> Today's Topics:
>
> 1. Fwd: When i do Video call from sipml5 to sipml5, Call get
> rejected (Anant Saraswat)
> 2. Re: Fwd: When i do Video call from sipml5 to sipml5, Call get
> rejected (sudhir mor)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 24 Oct 2013 20:41:40 +0530
> From: Anant Saraswat <anant.saraswat at techblue.co.uk>
> To: asterisk-video at lists.digium.com
> Subject: [Asterisk-video] Fwd: When i do Video call from sipml5 to
> sipml5, Call get rejected
> Message-ID: <526938AC.8000500 at techblue.co.uk>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>
> 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
>
>
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Fri, 25 Oct 2013 00:01:49 +0800 (SGT)
> From: sudhir mor <sudhir_mor2000 at yahoo.com>
> To: Development discussion of video media support in Asterisk
> <asterisk-video at lists.digium.com>
> Subject: Re: [Asterisk-video] Fwd: When i do Video call from sipml5 to
> sipml5, Call get rejected
> Message-ID:
> <1382630509.9970.YahooMailNeo at web190506.mail.sg3.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Use asterisk 1.4.15 with sergio mullia pkg.
>
> Regards?
> Sudhir Mor
> MaiBiz Technologies Private Limited
> RZ-6 GopalNagar near Khohwal Dharam Kanta,?
> Dhansa Road, Najafgarh, New Delhi.
> PIN - 110043?
> Mob: +91 - 9891318796,
> Email:?s at maibiz.in
> Skype: sudhirmor1
>
> ________________________________
>
>
>
>
> On Thursday, 24 October 2013 8:42 PM, Anant Saraswat <
> anant.saraswat at techblue.co.uk> wrote:
>
>
> 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
>
>
>
>
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