<div>Hi</div>
<div>I went through the log and it says "error opening encoder". I am using the following libraries<br> >>ldd app_transcoder.so <br> linux-gate.so.1 => (0xb8091000)<br> libavcodec.so.52 => /usr/lib/i686/cmov/libavcodec.so.52 (0xb78a6000)<br>
libswscale.so.0 => /usr/lib/i686/cmov/libswscale.so.0 (0xb7876000)<br> libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0xb7712000)<br> libavutil.so.49 => /usr/lib/i686/cmov/libavutil.so.49 (0xb7700000)<br> libz.so.1 => /lib/libz.so.1 (0xb76ea000)<br>
libm.so.6 => /lib/tls/i686/cmov/libm.so.6 (0xb76c4000)<br> libgsm.so.1 => /usr/lib/libgsm.so.1 (0xb76b7000)<br> libschroedinger-1.0.so.0 => /usr/lib/libschroedinger-1.0.so.0 (0xb7646000)<br> libpthread.so.0 => /lib/tls/i686/cmov/libpthread.so.0 (0xb762d000)<br>
libspeex.so.1 => /usr/lib/sse2/libspeex.so.1 (0xb7610000)<br> libtheora.so.0 => /usr/lib/libtheora.so.0 (0xb75bf000)<br> libvorbisenc.so.2 => /usr/lib/libvorbisenc.so.2 (0xb74c5000)<br> libvorbis.so.0 => /usr/lib/libvorbis.so.0 (0xb749b000)<br>
/lib/ld-linux.so.2 (0xb8092000)<br> liboil-0.3.so.0 => /usr/lib/liboil-0.3.so.0 (0xb742b000)<br> libogg.so.0 => /usr/lib/libogg.so.0 (0xb7425000)<br> librt.so.1 => /lib/tls/i686/cmov/librt.so.1 (0xb741c000)<br>
<br></div>
<div> </div>
<div><br>[Nov 2 02:39:59] WARNING[27345] app_transcoder.c: >Transcoding [,102@default,h2<br><a href="mailto:63@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50" target="_blank">63@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50</a>]<br>
[Nov 2 02:39:59] WARNING[27345] app_transcoder.c: >anand-Transcoding [,102@defa<br>
ult,h263@qcif/fps=10/kb=52/qmin=4/qmax=12/gs=50]<br>[Nov 2 02:39:59] WARNING[27345] app_transcoder.c: -Transcoder [f=0,fps=10,kb=53<br>248,qmin=4,qmax=12,gs=50]<br>[[Nov 2 02:39:59] ERROR[27345] app_transcoder.c: Error opening encoder<br>
[Nov 2 02:39:59] WARNING[27345] app_transcoder.c: -joining thread<br>[Nov 2 02:39:59] WARNING[27345] app_transcoder.c: -joined thread<br>[Nov 2 02:40:38] WARNING[27345] app_transcoder.c: -end loop[Nov 2 02:40:38] WA<br>
RNING[27345] app_transcoder.c: -Hanging up<br>[Nov 2 02:40:38] WARNING[27345] app_transcoder.c: <Transcoding<br>[Nov 2 02:40:58] WARNING[27324] chan_sip.c: Maximum retries exceeded on transmi<br>ssion <a href="mailto:1f511c6a092c41572f06f0c0353bdbc9@192.168.1.3" target="_blank">1f511c6a092c41572f06f0c0353bdbc9@192.168.1.3</a> for seqno 103 (Non-critical R<br>
equest)</div>
<div> </div>
<div>Please help me out.<br>Thanks<br>Anand<br><br><br> </div>
<div><span class="gmail_quote">On 02/11/2009, <b class="gmail_sendername">anandadip mandal</b> <<a href="mailto:anandadip@gmail.com" target="_blank">anandadip@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div>Hi</div>
<div>I want to make video call between two sip phone having different video codecs using app_transcoder.</div>
<div>I have used the following dialplan</div>
<div>[default]<br>exten => 101,1,Answer<br>exten => 101,2,transcode(,102@default,h263@qcif/fps=12/kb=52/qmin=4/qmax=12/gs=50)<br>exten => 102,1,Dial(SIP/101)</div>
<div> </div>
<div>the 102 ( having h263-1998 codec) extension is calling 101 (having h263 codec).</div>
<div>I can see the call between the two phone established but no video; also i dont see any ack coming from 101 and within seconds asterisk gives a segfault.</div>
<div>Without app transcoder, video call works fine when both phone use h263-1998 codec.</div>
<div>I am using asterisk 1.4; the transcode module loads succesfully; even it executes and places a call to the configured extension)</div>
<div> </div>
<div>Please help me if i am using the correct dialplan or am i missing something.</div>
<div> </div>
<div>Any help will be much appreciated.</div>
<div> </div>
<div>Regards</div>
<div>Anand</div>
<div><span>
<div><br><br> </div>
<div><span class="gmail_quote">On 26/10/2009, <b class="gmail_sendername">anandadip mandal</b> <<a href="mailto:anandadip@gmail.com" target="_blank">anandadip@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div>Hi </div>
<div>I have successfully compiled and able to load the app_transcoder.so;</div>
<div>I want to know the configuration of extension.conf to put the app_transcoder in use.</div>
<div>I have two sip soft phone(video capable) 3000, 3001 which are already registered to asterisk and I can make audio call between them;</div>
<div>Also please let me know if i have to add anything specific to extesion.conf and sip.conf for enabling video call.</div>
<div>Any help will be very much appreciated.</div>
<div>Thanks and regards</div>
<div>Anand</div>
<div><br> </div>
<div class="gmail_quote">2009/10/20 anandadip mandal <span dir="ltr"><<a href="mailto:anandadip@gmail.com" target="_blank">anandadip@gmail.com</a>></span><span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div>is there any document for compilation procedure of app transcoder?also could someone point me how to integrate it with asterisk?</div>
<div>Thanks</div>
<div>Anand<br clear="all"></div>
<div></div><br></blockquote></span></div><br><br clear="all"><br>-- <br><span>Anandadip Mandal<br></span></blockquote></div><br><br clear="all"><br></span></div>-- <br><span>Anandadip Mandal </span></blockquote>
</div><br><br clear="all"><br>-- <br>Anandadip Mandal