<div dir="ltr"><br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Oct 24, 2013 at 10:30 PM, <span dir="ltr"><<a href="mailto:asterisk-video-request@lists.digium.com" target="_blank">asterisk-video-request@lists.digium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Send asterisk-video mailing list submissions to<br>
<a href="mailto:asterisk-video@lists.digium.com">asterisk-video@lists.digium.com</a><br>
<br>
To subscribe or unsubscribe via the World Wide Web, visit<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br>
or, via email, send a message with subject or body 'help' to<br>
<a href="mailto:asterisk-video-request@lists.digium.com">asterisk-video-request@lists.digium.com</a><br>
<br>
You can reach the person managing the list at<br>
<a href="mailto:asterisk-video-owner@lists.digium.com">asterisk-video-owner@lists.digium.com</a><br>
<br>
When replying, please edit your Subject line so it is more specific<br>
than "Re: Contents of asterisk-video digest..."<br>
<br>
<br>
Today's Topics:<br>
<br>
1. Fwd: When i do Video call from sipml5 to sipml5, Call get<br>
rejected (Anant Saraswat)<br>
2. Re: Fwd: When i do Video call from sipml5 to sipml5, Call get<br>
rejected (sudhir mor)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Thu, 24 Oct 2013 20:41:40 +0530<br>
From: Anant Saraswat <<a href="mailto:anant.saraswat@techblue.co.uk">anant.saraswat@techblue.co.uk</a>><br>
To: <a href="mailto:asterisk-video@lists.digium.com">asterisk-video@lists.digium.com</a><br>
Subject: [Asterisk-video] Fwd: When i do Video call from sipml5 to<br>
sipml5, Call get rejected<br>
Message-ID: <<a href="mailto:526938AC.8000500@techblue.co.uk">526938AC.8000500@techblue.co.uk</a>><br>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>
<br>
<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>
<a href="http://192.168.100.71:42822" target="_blank">192.168.100.71:42822</a><br>
-- SIP/1060-00000001 is busy<br>
== Everyone is busy/congested at this time (2:1/0/1)<br>
-- Executing [1060@default:50006] Goto("SIP/1061-00000000",<br>
"stdexten-BUSY,1") in new stack<br>
-- Goto (default,stdexten-BUSY,1)<br>
-- Executing [stdexten-BUSY@default: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 href="http://2a8263684cfc957e7da826920c0e59cb@192.168.100.160:5060" target="_blank">2a8263684cfc957e7da826920c0e59cb@192.168.100.160:5060</a>'. 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 '<a href="http://192.168.100.71:42822" target="_blank">192.168.100.71:42822</a>' closed<br>
<br>
<br>
<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 2<br>
Date: Fri, 25 Oct 2013 00:01:49 +0800 (SGT)<br>
From: sudhir mor <<a href="mailto:sudhir_mor2000@yahoo.com">sudhir_mor2000@yahoo.com</a>><br>
To: Development discussion of video media support in Asterisk<br>
<<a href="mailto:asterisk-video@lists.digium.com">asterisk-video@lists.digium.com</a>><br>
Subject: Re: [Asterisk-video] Fwd: When i do Video call from sipml5 to<br>
sipml5, Call get rejected<br>
Message-ID:<br>
<<a href="mailto:1382630509.9970.YahooMailNeo@web190506.mail.sg3.yahoo.com">1382630509.9970.YahooMailNeo@web190506.mail.sg3.yahoo.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Use asterisk 1.4.15 with sergio mullia pkg.<br>
<br>
Regards?<br>
Sudhir Mor<br>
MaiBiz Technologies Private Limited<br>
RZ-6 GopalNagar near Khohwal Dharam Kanta,?<br>
Dhansa Road, Najafgarh, New Delhi.<br>
PIN - 110043?<br>
Mob: +91 - 9891318796,<br>
Email:?<a href="mailto:s@maibiz.in">s@maibiz.in</a><br>
Skype: sudhirmor1<br>
<br>
________________________________<br>
<br>
<br>
<br>
<br>
On Thursday, 24 October 2013 8:42 PM, Anant Saraswat <<a href="mailto:anant.saraswat@techblue.co.uk">anant.saraswat@techblue.co.uk</a>> wrote:<br>
<br>
<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>
<a href="http://192.168.100.71:42822" target="_blank">192.168.100.71:42822</a><br>
? ? -- SIP/1060-00000001 is busy<br>
? == Everyone is busy/congested at this time (2:1/0/1)<br>
? ? -- Executing [1060@default:50006] Goto("SIP/1061-00000000",<br>
"stdexten-BUSY,1") in new stack<br>
? ? -- Goto (default,stdexten-BUSY,1)<br>
? ? -- Executing [stdexten-BUSY@default: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 href="http://2a8263684cfc957e7da826920c0e59cb@192.168.100.160:5060" target="_blank">2a8263684cfc957e7da826920c0e59cb@192.168.100.160:5060</a>'. 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 '<a href="http://192.168.100.71:42822" target="_blank">192.168.100.71:42822</a>' 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>
-------------- next part --------------<br>
An HTML attachment was scrubbed...<br>
URL: <<a href="http://lists.digium.com/pipermail/asterisk-video/attachments/20131025/fcf05372/attachment-0001.html" target="_blank">http://lists.digium.com/pipermail/asterisk-video/attachments/20131025/fcf05372/attachment-0001.html</a>><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>
End of asterisk-video Digest, Vol 84, Issue 1<br>
*********************************************<br>
</blockquote></div><br><br clear="all"><br>-- <br><div dir="ltr"><div><div><div>IT DESK <br></div><a href="http://www.essencekey.com" target="_blank">www.essencekey.com</a><br></div><a href="mailto:info@essencekey.com" target="_blank">info@essencekey.com</a><br>
</div>Contact: +91-1164722856</div>
</div>