<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div><span>Use asterisk 1.4.15 with sergio mullia pkg.</span></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal;"><span><br></span></div><div></div><div>Regards </div><div><b><font color="#0000bf">Sudhir Mor</font></b></div><div><div class="im" style="color: rgb(80, 0, 80); font-family: arial, sans-serif; line-height: normal;"><div>MaiBiz Technologies Private Limited</div><div>RZ-6 GopalNagar near Khohwal Dharam Kanta, </div><div>Dhansa Road, Najafgarh, New Delhi.</div><div>PIN - 110043 </div><div>Mob: +91 - 9891318796,</div></div><div style="color: rgb(34, 34, 34); font-family: arial, sans-serif; line-height: normal;">Email: <a rel="nofollow" target="_blank" href="mailto:s@maibiz.in"
style="color:rgb(17, 85, 204);">s@maibiz.in</a></div></div><div style="color: rgb(80, 0, 80); font-family: arial, sans-serif; line-height: normal; font-size: 13px; background-color: transparent; font-style: normal;">Skype: sudhirmor1</div><div><b><font color="#00ffff"><hr></font></b></div><div class="yahoo_quoted" style="display: block;"> <br> <br> <div style="font-family: 'times new roman', 'new york', times, serif; font-size: 12pt;"> <div style="font-family: 'times new roman', 'new york', times, serif; font-size: 12pt;"> <div dir="ltr"> <font size="2" face="Arial"> On Thursday, 24 October 2013 8:42 PM, Anant Saraswat <anant.saraswat@techblue.co.uk> wrote:<br> </font> </div> <div class="y_msg_container"><br>Hello All,<br><br>I am using Asterisk 12 and sipml5 as front-end and when i call from one<br>to another the call will ring on other end but when i allow the camera<br>access call will terminated automatically. I have attached the logs
of<br>Asterisk, if some one will get something useful Please reply on the same.<br><br><br>Thanks and Regards,<br>Anant<br><br><br><br> == Using SIP VIDEO CoS mark 6<br> == Using SIP RTP CoS mark 5<br>[Oct 24 19:45:59] ERROR[3005][C-00000000]: netsock2.c:269<br>ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)",<br>...): Name or service not known<br>[Oct 24 19:45:59] WARNING[3005][C-00000000]: chan_sip.c:16067<br>__set_address_from_contact: Invalid host name in Contact: (can't resolve<br>in DNS) : 'df7jal23ls0d.invalid'<br>[Oct 24 19:45:59] ERROR[3005][C-00000000]: netsock2.c:98<br>ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported<br>[Oct 24 19:45:59] WARNING[3005][C-00000000]: app_dial.c:2423<br>dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 -<br>Subscriber absent)<br> -- Called SIP/1060<br> -- SIP/1060-00000001 is ringing<br> -- Got SIP response
603 "Failed to get local SDP" back from<br>192.168.100.71:42822<br> -- SIP/1060-00000001 is busy<br> == Everyone is busy/congested at this time (2:1/0/1)<br> -- Executing [<a ymailto="mailto:1060@default" href="mailto:1060@default">1060@default</a>:50006] Goto("SIP/1061-00000000",<br>"stdexten-BUSY,1") in new stack<br> -- Goto (default,stdexten-BUSY,1)<br> -- Executing [<a ymailto="mailto:stdexten-BUSY@default" href="mailto:stdexten-BUSY@default">stdexten-BUSY@default</a>:1]<br>VoiceMail("SIP/1061-00000000", "1060,b") in new stack<br>[Oct 24 19:46:07] WARNING[3003][C-00000000]: chan_sip.c:24402<br>handle_response: Remote host can't match request ACK to call<br>'<a ymailto="mailto:2a8263684cfc957e7da826920c0e59cb@192.168.100.160" href="mailto:2a8263684cfc957e7da826920c0e59cb@192.168.100.160">2a8263684cfc957e7da826920c0e59cb@192.168.100.160</a>:5060'. Giving up.<br> --
<SIP/1061-00000000> Playing 'vm-theperson.gsm' (language 'en')<br> -- <SIP/1061-00000000> Playing 'digits/1.gsm' (language 'en')<br> -- <SIP/1061-00000000> Playing 'digits/0.gsm' (language 'en')<br> -- <SIP/1061-00000000> Playing 'digits/6.gsm' (language 'en')<br> -- <SIP/1061-00000000> Playing 'digits/0.gsm' (language 'en')<br> -- <SIP/1061-00000000> Playing 'vm-isonphone.gsm' (language 'en')<br> -- <SIP/1061-00000000> Playing 'vm-intro.gsm' (language 'en')<br> -- <SIP/1061-00000000> Playing 'beep.gsm' (language 'en')<br> -- Recording the message<br> -- x=0, open writing:<br>/var/spool/asterisk/voicemail/default/1060/tmp/tJ2W4E format: wav49,<br>0x7fb880008408<br> -- x=1, open writing:<br>/var/spool/asterisk/voicemail/default/1060/tmp/tJ2W4E format:
gsm,<br>0x7fb88000f618<br> -- x=2, open writing:<br>/var/spool/asterisk/voicemail/default/1060/tmp/tJ2W4E format: wav,<br>0x7fb8800244d8<br>[Oct 24 19:46:23] WARNING[3005][C-00000000]: app.c:1384<br>__ast_play_and_record: No audio available on SIP/1061-00000000??<br> -- User hung up<br> == Spawn extension (default, stdexten-BUSY, 1) exited non-zero on<br>'SIP/1061-00000000'<br> == WebSocket connection from '192.168.100.71:42822' closed<br><br><br><br><br>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com </a>--<br><br>asterisk-video mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br><br><br></div> </div>
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