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Hello, I have been trying to playback a video file via Playback() in Asterisk 1.8.7.1 but the file format seems to fail. [2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format [2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4 (ulaw)): No such file or directory The file of course exists and it's chowned to asterisk.asterisk. I think it's a fil...

...map:98 H264/90000 a=fmtp:98 profile-level-id=428016;max-mbps=35000;max-fs=3600;max-smbps=323500 a=rtpmap:99 H263-1998/90000 a=fmtp:99 custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo a=rtpmap:34 H263/90000 a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200 a=rtpmap:31 H261/90000 a=fmtp:31 cif=1;qcif=1;maxbr=19200 a=rtcp-fb:* nack pli a=sendrecv a=content:main a=label:11 a=answer:full m=application 5078 UDP/BFCP * c=IN IP4 192.168.50.10 a=floorctrl:c-s a=confid:1 a=floorid:2 mstrm:12 a=userid:1 a=setup:passive a=connection:new m=...

We are experiencing an issue with our Cisco 9971 and 8945 phones where H264 video calls are connecting at 176x144 resolution instead of 640x480. Soft clients can connect at higher resolutions and the 9971 can even receive video at a higher resolution (although it still sends 176x144). I contacted one of the developers and he suggested the passthrough of SDP attributes is not working correctly.

...o 50946 RTP/AVP 8 2 18 3 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 35878 RTP/AVP 34 100 99 b=AS:128 a=sendrecv a=rtpmap:34 H263/90000 a=fmtp:34 CIF=2; QCIF=2 a=rtpmap:100 H263-1998/90000 a=fmtp:100 CIF=2; QCIF=2 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg== [Dec 6 15:11:18] Using INVITE request as basis request - 1666548288-45310-6 at BJC.BGI.B.BAD [Dec 6 15:11:18] Found...

...ils with attachments up to 200KB > Document viewer for MS-Office and PDF files Web browser: HTML4.01, JAVAScript1.5, SSL3.0, HTTP1.1, CSS1.0 Instant messaging: QQ Multimedia Features Video format: MP4, 3GPP Audio format: MP3, WAV, MIDI, AMR Picture format: WBMP, BMP, JPEG, GIF Camcorder: QVGA, QCIF Media Player > Audio: MP3 player > Video: up to 30 frames/second QVGA MP4/3GPP PIM Features Calendar Schedule management Alarm clock Voice recorder World time Currency converter Anniversary Other Features English <-> Chinese dictionary Calculator World time Notepad Sketch pad File t...

...v=0 o=- 9 2 IN IP4 172.16.148.129 s=CounterPath eyeBeam 1.5 c=IN IP4 172.16.148.129 t=0 0 m=audio 18342 RTP/AVP 18 3 0 101 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:07F885091DAF4305868C0F432F6512CF m=video 35840 RTP/AVP 34 115 125 a=fmtp:34 QCIF=2 MAXBR=1960 a=fmtp:115 QCIF=2 MAXBR=1960 a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:125 H264/90000 a=sendrecv a=x-rtp-session-id:399907E52958448ABF8B7176DCD44BD5 <-------------> --- (11 headers 20 lines) --- F...

...Video Phones, and i see that Asterisk is stripping the fmtp parameters for the h263 video line in SDP.<br>For example a line similar to the below is stripped,<br> <br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp; a=fmtp:xx CIF=4;QCIF=2;F=1;K=1<br><br>Asterisk is configured NOT to be present in the Media path (My version : Asterisk <a href="http://1.4.19.1">1.4.19.1</a> ).<br>I have the following enabled in my sip.conf.<br> <br>canreinvite=yes <br>directrtpsetup=yes <b...

Hello All, I'am doing a video call between two Video Phones, and i see that Asterisk is stripping the fmtp parameters for the h263 video line in SDP. For example a line similar to the below is stripped, a=fmtp:xx CIF=4;QCIF=2;F=1;K=1 Asterisk is configured NOT to be present in the Media path (My version : Asterisk 1.4.19.1 ). I have the following enabled in my sip.conf. canreinvite=yes directrtpsetup=yes

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

How good is the video compression in vp3.2? For example, how does it compare (bit rate & quality) to mpeg1, mpeg2, Divx, real Video 8+, Windows Media, etc. etc. It being open source is nice and all, but if it can't be reasonably competitive then there isn't much point to it. I just haven't seen anything that compares vp32 to anything else. (Although to be honest, I haven't

How does the Theora project differ from Nullsofts NSV project? Other than the obvious fact they use mp3.... <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'theora-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic