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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Two items<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>#1 you only need 1 disallow=all in your sip.conf definition<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>#2 you need to patch rtp.c to define 126 as FORMAT_H263 – this is an xlite response to Asterisk starting music-on-hold during the connect pause. The r on the dial command attempts to do a “faux ring” which xlite interprets as a MOH request, so if you don’t want to patch/recompile, just take the r off of Dial.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> virendra bhati [mailto:virbhati@gmail.com] <br><b>Sent:</b> Monday, November 21, 2011 4:35 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion; Sam Govind; Danny Nicholas<br><b>Subject:</b> video calls not working<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Hi list,<b><br><br></b>I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.<b><br><br>Extensions.conf</b><br><br>exten => 111,1,Answer()<br> same => n,Dial(SIP/2206,60,r)<br> same => n,Hangup()<br><br><b>SIP.conf</b><br>[2218]<br><br>type=friend<br>secret=*******<br>callerid="Virendra" <9172341457><br>host=dynamic ; This device needs to register<br>nat=yes ; X-Lite is behind a NAT router<br>canreinvite=no ; Typically set to NO if behind NAT<br>disallow=all<br>dtmfmode=inband<br>insecure=invite,port<br>;context=outbound<br>context=bhati-test<br>qualify=yes<br>accountcode=123654789<br>disallow = all<br>allow = ulaw,alaw,h263,g729,gsm,h264<br>videosupport=yes<br><br>[2206]<br>type=friend<br>secret=*******<br>callerid=2206<br>host=dynamic ; This device needs to register<br>nat=yes ; X-Lite is behind a NAT router<br>canreinvite=no ; Typically set to NO if behind NAT<br>disallow=all<br>dtmfmode=inband<br>insecure=invite,port<br>context=outbound<br>qualify=yes<br>disallow = all<br>allow = ulaw,alaw,h263,g729,gsm,h264<br>videosupport=yes<br><br><b>codec list of asterisk 1.6.2.11</b><br><br><b>haddock8-astrx*CLI> core show codecs</b><br>Disclaimer: this command is for informational purposes only.<br> It does not indicate anything about your configuration.<br> INT BINARY HEX TYPE NAME DESC<br>--------------------------------------------------------------------------------<br> 1 (1 << 0) (0x1) audio g723 (G.723.1)<br> 2 (1 << 1) (0x2) audio gsm (GSM)<br> 4 (1 << 2) (0x4) audio ulaw (G.711 u-law)<br> 8 (1 << 3) (0x8) audio alaw (G.711 A-law)<br> 16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2)<br> 32 (1 << 5) (0x20) audio adpcm (ADPCM)<br> 64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM)<br> 128 (1 << 7) (0x80) audio lpc10 (LPC10)<br> 256 (1 << 8) (0x100) audio g729 (G.729A)<br> 512 (1 << 9) (0x200) audio speex (SpeeX)<br> 1024 (1 << 10) (0x400) audio ilbc (iLBC)<br> 2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551)<br> 4096 (1 << 12) (0x1000) audio g722 (G722)<br> 65536 (1 << 16) (0x10000) image jpeg (JPEG image)<br> 131072 (1 << 17) (0x20000) image png (PNG image)<br> 262144 (1 << 18) (0x40000) video h261 (H.261 Video)<br> 524288 (1 << 19) (0x80000) video h263 (H.263 Video)<br> 1048576 (1 << 20) (0x100000) video h263p (H.263+ Video)<br> 2097152 (1 << 21) (0x200000) video h264 (H.264 Video)<br>haddock8-astrx*CLI><br><br><br><b>CLI Output:-</b><br><br clear=all><span style='color:#660000'> -- Executing [111@bhati-test:1] Answer("SIP/2218-00000664", "") in new stack<br> -- Executing [111@bhati-test:2] Dial("SIP/2218-00000664", "SIP/2206,60,r") in new stack<br> == Using SIP RTP CoS mark 5<br> == Using SIP VRTP CoS mark 6<br> -- Called 2206<br> -- SIP/2206-00000665 is ringing<br> -- SIP/2206-00000665 is ringing<br>[Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br>[Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br>[Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br> -- SIP/2206-00000665 answered SIP/2218-00000664<br>[Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191'<br>[Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191'<br>[Nov 21 15:57:24] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191'<br>[Nov 21 15:57:30] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br>[Nov 21 15:57:34] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191'<br>[Nov 21 15:57:40] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br>[Nov 21 15:57:44] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191'<br>[Nov 21 15:57:50] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br>[Nov 21 15:57:54] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191'<br>[Nov 21 15:58:00] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br>[Nov 21 15:58:04] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191'<br>[Nov 21 15:58:11] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br>[Nov 21 15:58:13] NOTICE[7924]: chan_sip.c:21479 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 2206<br>[Nov 21 15:58:15] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191'<br>[Nov 21 15:58:21] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br>[Nov 21 15:58:25] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191'<br>[Nov 21 15:58:30] NOTICE[7924]: chan_sip.c:21479 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 2218<br>[Nov 21 15:58:31] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br>[Nov 21 15:58:35] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.191'<br>[Nov 21 15:58:41] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40'<br> -- Executing [h@bhati-test:1] NoOp("SIP/2218-00000664", "hangup the call now") in new stack<br> == Spawn extension (bhati-test, 111, 2) exited non-zero on 'SIP/2218-00000664'<br></span><br>-- <o:p></o:p></p><div><p class=MsoNormal><br><br><br>-----<br>Thanks and regards<br><br> Virendra Bhati<br>+91-9172341457<br>Software Engineer<o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>