On Thu, Oct 24, 2013 at 10:30 PM, <[email protected]>wrote:
> Send asterisk-video mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-video > or, via email, send a message with subject or body 'help' to > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-video digest..." > > > 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 <[email protected]> > To: [email protected] > Subject: [Asterisk-video] Fwd: When i do Video call from sipml5 to > sipml5, Call get rejected > Message-ID: <[email protected]> > 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@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 > '[email protected]: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 <[email protected]> > To: Development discussion of video media support in Asterisk > <[email protected]> > Subject: Re: [Asterisk-video] Fwd: When i do Video call from sipml5 to > sipml5, Call get rejected > Message-ID: > <[email protected]> > 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:[email protected] > Skype: sudhirmor1 > > ________________________________ > > > > > On Thursday, 24 October 2013 8:42 PM, Anant Saraswat < > [email protected]> 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@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 > '[email protected]: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 -- > > asterisk-video mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-video > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.digium.com/pipermail/asterisk-video/attachments/20131025/fcf05372/attachment-0001.html > > > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-video mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-video > > End of asterisk-video Digest, Vol 84, Issue 1 > ********************************************* > -- IT DESK www.essencekey.com [email protected] Contact: +91-1164722856
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